chan_sip.c File Reference

Implementation of Session Initiation Protocol. More...

#include "asterisk.h"
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <inttypes.h>
#include "asterisk/network.h"
#include "asterisk/paths.h"
#include "asterisk/lock.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock2.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/cel.h"
#include "asterisk/data.h"
#include "asterisk/aoc.h"
#include "sip/include/sip.h"
#include "sip/include/globals.h"
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
#include "sip/include/srtp.h"
#include "sip/include/sdp_crypto.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"

Go to the source code of this file.

Data Structures
struct	ast_register_list
	The register list: Other SIP proxies we register with and receive calls from. More...
struct	ast_subscription_mwi_list
	The MWI subscription list. More...
struct	cfsip_methods
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type. More...
struct	cfsubscription_types
	Subscription types that we support. We support dialoginfo updates (really device status, not dialog info as was the original intent of the standard) SIMPLE presence used for device status Voicemail notification subscriptions. More...
struct	domain_list
struct	epa_static_data_list
struct	event_state_compositor
	The Event State Compositors. More...
struct	invstate2stringtable
	Readable descriptions of device states. More...
struct	match_req_args
struct	sip_history_head
struct	sip_reasons
	Diversion header reasons. More...
Object counters @{
Bug: These counters are not handled in a thread-safe way ast_atomic_fetchadd_int() should be used to modify these values.
#define	append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)
	Append to SIP dialog history.
#define	CHECK_AUTH_BUF_INITLEN   256
#define	check_request_transport(peer, tmpl)
	generic function for determining if a correct transport is being used to contact a peer
#define	DATA_EXPORT_SIP_PEER(MEMBER)
#define	FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"
#define	FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"
#define	FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"
#define	FORMAT   "%-39.39s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
#define	FORMAT   "%-40.40s %-20.20s %-16.16s\n"
#define	FORMAT   "%-25.25s %-39.39s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
#define	FORMAT   "%-47.47s %-9.9s %-6.6s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-10.10s %-10.10s\n"
#define	FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define	FORMAT2   "%-39.39s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define	FORMAT2   "%-25.25s %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-10s %s\n"
#define	FORMAT2   "%-47.47s %9.9s %6.6s\n"
#define	FORMAT2   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
#define	FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
#define	sip_pvt_lock(x)   ao2_lock(x)
#define	sip_pvt_trylock(x)   ao2_trylock(x)
#define	sip_pvt_unlock(x)   ao2_unlock(x)
#define	UNLINK(element, head, prev)
enum	match_req_res { SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, SIP_REQ_LOOP_DETECTED }
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .reload = reload, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .nonoptreq = "res_crypto,chan_local", }
static int	apeerobjs = 0
static char *	app_dtmfmode = "SIPDtmfMode"
static char *	app_sipaddheader = "SIPAddHeader"
static char *	app_sipremoveheader = "SIPRemoveHeader"
static struct ast_module_info *	ast_module_info = &__mod_info
static struct sip_auth *	authl = NULL
	Authentication list for realm authentication.
struct ast_sockaddr	bindaddr
static struct epa_static_data	cc_epa_static_data
static struct ast_threadstorage	check_auth_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_check_auth_buf , .custom_init = NULL , }
static struct ast_custom_function	checksipdomain_function
static struct ast_cli_entry	cli_sip []
	SIP Cli commands definition.
static struct ast_sockaddr	debugaddr
static const int	DEFAULT_PUBLISH_EXPIRES = 3600
static struct ast_tls_config	default_tls_cfg
	Default TLS connection configuration.
static struct ao2_container *	dialogs
	Here we implement the container for dialogs (sip_pvt), defining generic wrapper functions to ease the transition from the current implementation (a single linked list) to a different container. In addition to a reference to the container, we need functions to lock/unlock the container and individual items, and functions to add/remove references to the individual items.
static struct _map_x_s	dtmfstr []
	mapping between dtmf flags and strings
static int	esc_etag_counter
static const int	ESC_MAX_BUCKETS = 37
static struct event_state_compositor	event_state_compositors []
	The Event State Compositors.
static struct ast_sockaddr	externaddr
	our external IP address/port for SIP sessions. externaddr.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:
static time_t	externexpire
static char	externhost [MAXHOSTNAMELEN]
static int	externrefresh = 10
static uint16_t	externtcpport
static uint16_t	externtlsport
static struct _map_x_s	faxecmodes []
static struct ast_flags	global_flags [3] = {{0}}
static int	global_t38_maxdatagram
static const int	HASH_DIALOG_SIZE = 563
static const int	HASH_PEER_SIZE = 563
static struct _map_x_s	insecurestr []
static struct ast_sockaddr	internip
	our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.
static struct io_context *	io
static struct ast_ha *	localaddr
	List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.
static struct ast_sockaddr	media_address
static pthread_t	monitor_thread = AST_PTHREADT_NULL
	This is the thread for the monitor which checks for input on the channels which are not currently in use.
static ast_mutex_t	monlock = { { { NULL }, { 0 }, 0, { NULL }, { 0 }, PTHREAD_MUTEX_INITIALIZER } , 1, PTHREAD_MUTEX_INITIALIZER }
	Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.
static ast_mutex_t	netlock = { { { NULL }, { 0 }, 0, { NULL }, { 0 }, PTHREAD_MUTEX_INITIALIZER } , 1, PTHREAD_MUTEX_INITIALIZER }
static int	network_change_event_sched_id = -1
static struct ast_event_sub *	network_change_event_subscription
static struct ast_config *	notify_types = NULL
static int	ourport_tcp
static int	ourport_tls
static struct ao2_container *	peers
	The peer list: Users, Peers and Friends.
static struct ao2_container *	peers_by_ip
static struct ast_data_handler	peers_data_provider
static struct _map_x_s	referstatusstrings []
static struct ast_register_list	regl
	The register list: Other SIP proxies we register with and receive calls from.
static int	regobjs = 0
static struct _map_x_s	regstatestrings []
static int	rpeerobjs = 0
struct sched_context *	sched
static struct ast_cc_agent_callbacks	sip_cc_agent_callbacks
static struct ast_cc_monitor_callbacks	sip_cc_monitor_callbacks
struct {
enum sip_cc_notify_state   state
const char *   state_string
}	sip_cc_notify_state_map []
struct {
enum ast_cc_service_type   service
const char *   service_string
}	sip_cc_service_map []
static struct ast_data_entry	sip_data_providers []
static struct ast_custom_function	sip_header_function
struct ao2_container *	sip_monitor_instances
static ast_mutex_t	sip_reload_lock = { { { NULL }, { 0 }, 0, { NULL }, { 0 }, PTHREAD_MUTEX_INITIALIZER } , 1, PTHREAD_MUTEX_INITIALIZER }
static int	sip_reloading = FALSE
static enum channelreloadreason	sip_reloadreason
static struct ast_rtp_glue	sip_rtp_glue
static struct ast_tcptls_session_args	sip_tcp_desc
	The TCP server definition.
struct ast_channel_tech	sip_tech
	Definition of this channel for PBX channel registration.
struct ast_channel_tech	sip_tech_info
	This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.
static struct ast_tls_config	sip_tls_cfg
	Working TLS connection configuration.
static struct ast_tcptls_session_args	sip_tls_desc
	The TCP/TLS server definition.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.
static struct ast_custom_function	sipchaninfo_function
	Structure to declare a dialplan function: SIPCHANINFO.
static enum sip_debug_e	sipdebug
static int	sipdebug_text
	extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.
static struct ast_custom_function	sippeer_function
	Structure to declare a dialplan function: SIPPEER.
static int	sipsock = -1
	Main socket for UDP SIP communication.
static int *	sipsock_read_id
static int	speerobjs = 0
static struct _map_x_s	stmodes []
	Report Peer status in character string.
static struct _map_x_s	strefreshers []
static struct ast_subscription_mwi_list	submwil
	The MWI subscription list.
static struct ao2_container *	threadt
	The table of TCP threads.
static struct ast_threadstorage	ts_temp_pvt = { .once = PTHREAD_ONCE_INIT , .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , }
static char	used_context [AST_MAX_CONTEXT]
static const char *	__get_header (const struct sip_request *req, const char *name, int *start)
static void	__init_check_auth_buf (void)
static void	__init_ts_temp_pvt (void)
	A per-thread temporary pvt structure.
static void	__reg_module (void)
static int	__set_address_from_contact (const char *fullcontact, struct ast_sockaddr *addr, int tcp)
int	__sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acknowledges receipt of a packet and stops retransmission called with p locked.
static int	__sip_autodestruct (const void *data)
	Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().
void	__sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)
	Execute destruction of SIP dialog structure, release memory.
static int	__sip_do_register (struct sip_registry *r)
	Register with SIP proxy.
void	__sip_pretend_ack (struct sip_pvt *p)
	Pretend to ack all packets called with p locked.
static int	__sip_reliable_xmit (struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
	Transmit packet with retransmits.
int	__sip_semi_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)
	Acks receipt of packet, keep it around (used for provisional responses).
static int	__sip_subscribe_mwi_do (struct sip_subscription_mwi *mwi)
	Actually setup an MWI subscription or resubscribe.
static int	__sip_xmit (struct sip_pvt *p, struct ast_str *data, int len)
	Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.
static int	__transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Base transmit response function.
static void	__unreg_module (void)
static char *	_sip_qualify_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Send qualify message to peer from cli or manager. Mostly for debugging.
static char *	_sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Show one peer in detail (main function).
static char *	_sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Execute sip show peers command.
static void *	_sip_tcp_helper_thread (struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
	SIP TCP thread management function This function reads from the socket, parses the packet into a request.
static void	add_blank (struct sip_request *req)
	add a blank line if no body
static void	add_cc_call_info_to_response (struct sip_pvt *p, struct sip_request *resp)
static void	add_codec_to_sdp (const struct sip_pvt *p, format_t codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_content (struct sip_request *req, const char *line)
	Add content (not header) to SIP message.
static int	add_digit (struct sip_request *req, char digit, unsigned int duration, int mode)
	Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.
static void	add_diversion_header (struct sip_request *req, struct sip_pvt *pvt)
	Add "Diversion" header to outgoing message.
static int	add_header (struct sip_request *req, const char *var, const char *value)
	Add header to SIP message.
static int	add_header_max_forwards (struct sip_pvt *dialog, struct sip_request *req)
	Add 'Max-Forwards' header to SIP message.
static void	add_noncodec_to_sdp (const struct sip_pvt *p, int format, struct ast_str **m_buf, struct ast_str **a_buf, int debug)
	Add RFC 2833 DTMF offer to SDP.
static void	add_peer_mailboxes (struct sip_peer *peer, const char *value)
static void	add_peer_mwi_subs (struct sip_peer *peer)
static struct sip_auth *	add_realm_authentication (struct sip_auth *authlist, const char *configuration, int lineno)
	Add realm authentication in list.
static void	add_route (struct sip_request *req, struct sip_route *route)
	Add route header into request per learned route.
static int	add_rpid (struct sip_request *req, struct sip_pvt *p)
	Add Remote-Party-ID header to SIP message.
static enum sip_result	add_sdp (struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
	Add Session Description Protocol message.
static int	add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)
	Add SIP domain to list of domains we are responsible for.
static int	add_supported_header (struct sip_pvt *pvt, struct sip_request *req)
	Add "Supported" header to sip message. Since some options may be disabled in the config, the sip_pvt must be inspected to determine what is supported for this dialog.
static void	add_tcodec_to_sdp (const struct sip_pvt *p, int codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add text codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_text (struct sip_request *req, const char *text)
	Add text body to SIP message.
static struct ast_variable *	add_var (const char *buf, struct ast_variable *list)
	implement the setvar config line
static void	add_vcodec_to_sdp (const struct sip_pvt *p, format_t codec, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add video codec offer to SDP offer/answer body in INVITE or 200 OK.
static int	add_vidupdate (struct sip_request *req)
	add XML encoded media control with update
static void	append_date (struct sip_request *req)
	Append date to SIP message.
static void	append_history_full (struct sip_pvt *p, const char *fmt,...)
	Append to SIP dialog history with arg list.
static void	append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)
	Append to SIP dialog history with arg list.
static int	apply_directmedia_ha (struct sip_pvt *p, const char *op)
	AST_DATA_STRUCTURE (sip_peer, DATA_EXPORT_SIP_PEER)
static void	ast_quiet_chan (struct ast_channel *chan)
	Turn off generator data XXX Does this function belong in the SIP channel?
static void	ast_sip_ouraddrfor (const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p)
	NAT fix - decide which IP address to use for Asterisk server?
static int	ast_sockaddr_resolve_first (struct ast_sockaddr *addr, const char *name, int flag)
	Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.
static int	ast_sockaddr_resolve_first_af (struct ast_sockaddr *addr, const char *name, int flag, int family)
	Return the first entry from ast_sockaddr_resolve filtered by address family.
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static void	auth_headers (enum sip_auth_type code, char **header, char **respheader)
	return the request and response heade for a 401 or 407 code
static int	auto_congest (const void *arg)
	Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.
static void	build_callid_pvt (struct sip_pvt *pvt)
	Build SIP Call-ID value for a non-REGISTER transaction.
static void	build_callid_registry (struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain)
	Build SIP Call-ID value for a REGISTER transaction.
static void	build_contact (struct sip_pvt *p)
	Build contact header - the contact header we send out.
static struct sip_peer *	build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only)
	Build peer from configuration (file or realtime static/dynamic).
static int	build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)
	Build reply digest.
static void	build_route (struct sip_pvt *p, struct sip_request *req, int backwards)
	Build route list from Record-Route header.
static void	build_via (struct sip_pvt *p)
	Build a Via header for a request.
static int	cb_extensionstate (char *context, char *exten, int state, void *data)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static void	cc_epa_destructor (void *data)
static void	cc_handle_publish_error (struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry)
static void	change_redirecting_information (struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, struct ast_set_party_redirecting *update_redirecting, int set_call_forward)
	update redirecting information for a channel based on headers
static void	change_t38_state (struct sip_pvt *p, int state)
	Change the T38 state on a SIP dialog.
static enum check_auth_result	check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, const char *uri, enum xmittype reliable, int ignore)
	Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).
static enum check_auth_result	check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct ast_sockaddr *addr, struct sip_peer **authpeer, enum xmittype reliable, char *calleridname, char *uri2)
	Validate device authentication.
static void	check_pendings (struct sip_pvt *p)
	Check pending actions on SIP call.
static void	check_rtp_timeout (struct sip_pvt *dialog, time_t t)
	helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
static int	check_sip_domain (const char *domain, char *context, size_t len)
	check_sip_domain: Check if domain part of uri is local to our server
static int	check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr)
	Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.
static enum check_auth_result	check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr, struct sip_peer **authpeer)
	Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.
static void	check_via (struct sip_pvt *p, struct sip_request *req)
	check Via: header for hostname, port and rport request/answer
static attribute_unused void	check_via_response (struct sip_pvt *p, struct sip_request *req)
	check received= and rport= in a SIP response. If we get a response with received= and/or rport= in the Via: line, use them as 'p->ourip' (see RFC 3581 for rport, and RFC 3261 for received). Using these two fields SIP can produce the correct address and port in the SIP headers without the need for STUN. The address part is also reused for the media sessions. Note that ast_sip_ouraddrfor() still rewrites p->ourip if you specify externaddr/seternaddr/.
static void	cleanup_stale_contexts (char *new, char *old)
	Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.
static void	clear_peer_mailboxes (struct sip_peer *peer)
static int	clear_realm_authentication (struct sip_auth *authlist)
	Clear realm authentication list (at reload).
static void	clear_sip_domains (void)
	Clear our domain list (at reload).
static char *	complete_sip_peer (const char *word, int state, int flags2)
	Do completion on peer name.
static char *	complete_sip_registered_peer (const char *word, int state, int flags2)
	Do completion on registered peer name.
static char *	complete_sip_show_history (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show history' CLI.
static char *	complete_sip_show_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show peer' CLI.
static char *	complete_sip_show_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show user' CLI.
static char *	complete_sip_unregister (const char *line, const char *word, int pos, int state)
	Support routine for 'sip unregister' CLI.
static char *	complete_sip_user (const char *word, int state)
	Do completion on user name.
static char *	complete_sipch (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.
static char *	complete_sipnotify (const char *line, const char *word, int pos, int state)
	Support routine for 'sip notify' CLI.
static int	construct_pidf_body (enum sip_cc_publish_state state, char *pidf_body, size_t size, const char *presentity)
static int	copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy all headers from one request to another.
static int	copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy one header field from one request to another.
static void	copy_request (struct sip_request *dst, const struct sip_request *src)
	copy SIP request (mostly used to save request for responses)
static void	copy_socket_data (struct sip_socket *to_sock, const struct sip_socket *from_sock)
static struct ast_variable *	copy_vars (struct ast_variable *src)
	duplicate a list of channel variables,
static int	copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy SIP VIA Headers from the request to the response.
static int	create_addr (struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog, struct ast_sockaddr *remote_address)
	create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success
static int	create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)
	Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.
static struct sip_epa_entry *	create_epa_entry (const char *const event_package, const char *const destination)
static struct sip_esc_entry *	create_esc_entry (struct event_state_compositor *esc, struct sip_request *req, const int expires)
static void	create_new_sip_etag (struct sip_esc_entry *esc_entry, int is_linked)
static void	deinit_req (struct sip_request *req)
	Deinitialize SIP response/request.
static void	destroy_association (struct sip_peer *peer)
	Remove registration data from realtime database or AST/DB when registration expires.
static void	destroy_escs (void)
static void	destroy_mailbox (struct sip_mailbox *mailbox)
static int	determine_firstline_parts (struct sip_request *req)
	Parse first line of incoming SIP request.
static enum sip_publish_type	determine_sip_publish_type (struct sip_request *req, const char *const event, const char *const etag, const char *const expires, int *expires_int)
static int	dialog_cmp_cb (void *obj, void *arg, int flags)
static int	dialog_dump_func (void *userobj, void *arg, int flags)
static int	dialog_find_multiple (void *obj, void *arg, int flags)
static int	dialog_hash_cb (const void *obj, const int flags)
static int	dialog_initialize_rtp (struct sip_pvt *dialog)
	Initialize RTP portion of a dialog.
static int	dialog_needdestroy (void *dialogobj, void *arg, int flags)
	Match dialogs that need to be destroyed.
struct sip_pvt *	dialog_ref_debug (struct sip_pvt *p, char *tag, char *file, int line, const char *func)
void *	dialog_unlink_all (struct sip_pvt *dialog, int lockowner, int lockdialoglist)
	Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.
struct sip_pvt *	dialog_unref_debug (struct sip_pvt *p, char *tag, char *file, int line, const char *func)
static void	disable_dsp_detect (struct sip_pvt *p)
static int	do_magic_pickup (struct ast_channel *channel, const char *extension, const char *context)
static void *	do_monitor (void *data)
	The SIP monitoring thread.
static int	do_proxy_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
	Add authentication on outbound SIP packet.
static int	do_register_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
	Authenticate for outbound registration.
static void	do_setnat (struct sip_pvt *p)
	Set nat mode on the various data sockets.
static const char *	domain_mode_to_text (const enum domain_mode mode)
	Print domain mode to cli.
static const char *	dtmfmode2str (int mode)
	Convert DTMF mode to printable string.
static void	enable_dsp_detect (struct sip_pvt *p)
static int	esc_cmp_fn (void *obj, void *arg, int flags)
static void	esc_entry_destructor (void *obj)
static int	esc_hash_fn (const void *obj, const int flags)
static int	expire_register (const void *data)
	Expire registration of SIP peer.
static void	extract_uri (struct sip_pvt *p, struct sip_request *req)
	Check Contact: URI of SIP message.
static const char *	faxec2str (int faxec)
static int	finalize_content (struct sip_request *req)
	Add 'Content-Length' header and content to SIP message.
static const char *	find_alias (const char *name, const char *_default)
	Find compressed SIP alias.
static int	find_by_callid_helper (void *obj, void *arg, int flags)
static int	find_by_name (void *obj, void *arg, void *data, int flags)
static int	find_by_notify_uri_helper (void *obj, void *arg, int flags)
static int	find_by_subscribe_uri_helper (void *obj, void *arg, int flags)
static struct sip_pvt *	find_call (struct sip_request *req, struct ast_sockaddr *addr, const int intended_method)
	find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_incoming(), sipsock_read
static int	find_calling_channel (void *obj, void *arg, void *data, int flags)
	Find the channel that is causing the RINGING update.
const char *	find_closing_quote (const char *start, const char *lim)
	Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.
static struct sip_peer *	find_peer (const char *peer, struct ast_sockaddr *addr, int realtime, int which_objects, int devstate_only, int transport)
	Locate device by name or ip address.
static struct sip_auth *	find_realm_authentication (struct sip_auth *authlist, const char *realm)
	Find authentication for a specific realm.
static int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
static struct ast_cc_agent *	find_sip_cc_agent_by_notify_uri (const char *const uri)
static struct ast_cc_agent *	find_sip_cc_agent_by_original_callid (struct sip_pvt *pvt)
static struct ast_cc_agent *	find_sip_cc_agent_by_subscribe_uri (const char *const uri)
static int	find_sip_method (const char *msg)
	find_sip_method: Find SIP method from header
static int	find_sip_monitor_instance_by_subscription_pvt (void *obj, void *arg, int flags)
static int	find_sip_monitor_instance_by_suspension_entry (void *obj, void *arg, int flags)
static struct epa_static_data *	find_static_data (const char *const event_package)
static struct cfsubscription_types *	find_subscription_type (enum subscriptiontype subtype)
	Find subscription type in array.
static void	free_old_route (struct sip_route *route)
	Remove route from route list.
static int	func_check_sipdomain (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	Dial plan function to check if domain is local.
static int	func_header_read (struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
	Read SIP header (dialplan function).
static int	function_sipchaninfo_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPCHANINFO()} Dialplan function - reads sip channel data
static int	function_sippeer (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPPEER()} Dialplan function - reads peer data
static char *	generate_random_string (char *buf, size_t size)
	Generate 32 byte random string for callid's etc.
static char *	generate_uri (struct sip_pvt *pvt, char *buf, size_t size)
static int	get_address_family_filter (const struct ast_sockaddr *addr)
	Helper for dns resolution to filter by address family.
static int	get_also_info (struct sip_pvt *p, struct sip_request *oreq)
	Call transfer support (old way, deprecated by the IETF).
static char *	get_body (struct sip_request *req, char *name, char delimiter)
	Get a specific line from the message body.
static char *	get_body_by_line (const char *line, const char *name, int nameLen, char delimiter)
	Reads one line of SIP message body.
static int	get_cached_mwi (struct sip_peer *peer, int *new, int *old)
	Get cached MWI info.
static void	get_crypto_attrib (struct sip_srtp *srtp, const char **a_crypto)
static enum sip_get_dest_result	get_destination (struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id)
	Find out who the call is for. We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.
static int	get_domain (const char *str, char *domain, int len)
	Extract domain from SIP To/From header.
static struct event_state_compositor *	get_esc (const char *const event_package)
static struct sip_esc_entry *	get_esc_entry (const char *entity_tag, struct event_state_compositor *esc)
static const char *	get_header (const struct sip_request *req, const char *name)
	Get header from SIP request.
static struct ast_variable *	get_insecure_variable_from_config (struct ast_config *config)
static int	get_ip_and_port_from_sdp (struct sip_request *req, const enum media_type media, struct ast_sockaddr *addr)
static int	get_msg_text (char *buf, int len, struct sip_request *req, int addnewline)
	Get text out of a SIP MESSAGE packet.
static const char *	get_name_from_variable (struct ast_variable *var, const char *newpeername)
static void	get_our_media_address (struct sip_pvt *p, int needvideo, int needtext, struct ast_sockaddr *addr, struct ast_sockaddr *vaddr, struct ast_sockaddr *taddr, struct ast_sockaddr *dest, struct ast_sockaddr *vdest, struct ast_sockaddr *tdest)
	Set all IP media addresses for this call.
static int	get_pai (struct sip_pvt *p, struct sip_request *req)
	Parse the parts of the P-Asserted-Identity header on an incoming packet. Returns 1 if a valid header is found and it is different from the current caller id.
static int	get_rdnis (struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason)
	Get referring dnis.
static void	get_realm (struct sip_pvt *p, const struct sip_request *req)
	Choose realm based on From header and then To header or use globaly configured realm. Realm from From/To header should be listed among served domains in config file: domain=...
static int	get_refer_info (struct sip_pvt *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
static int	get_rpid (struct sip_pvt *p, struct sip_request *oreq)
	Get name, number and presentation from remote party id header, returns true if a valid header was found and it was different from the current caller id.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
static char	get_sdp_line (int *start, int stop, struct sip_request *req, const char **value)
	Fetches the next valid SDP line between the 'start' line (inclusive) and the 'stop' line (exclusive). Returns the type ('a', 'c', ...) and matching line in reference 'start' is updated with the next line number.
static struct sip_pvt *	get_sip_pvt_byid_locked (const char *callid, const char *totag, const char *fromtag)
	Lock dialog lock and find matching pvt lock.
static const char *	get_srv_protocol (enum sip_transport t)
	Return protocol string for srv dns query.
static const char *	get_srv_service (enum sip_transport t)
	Return service string for srv dns query.
static const char *	get_transport (enum sip_transport t)
	Return transport as string.
static const char *	get_transport_list (unsigned int transports)
	Return configuration of transports for a device.
static const char *	get_transport_pvt (struct sip_pvt *p)
	Return transport of dialog.
static int	get_transport_str2enum (const char *transport)
	Return int representing a bit field of transport types found in const char *transport.
static const char *	gettag (const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
	Get tag from packet.
static int	handle_cc_notify (struct sip_pvt *pvt, struct sip_request *req)
static int	handle_cc_subscribe (struct sip_pvt *p, struct sip_request *req)
static int	handle_common_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
	Handle flag-type options common to configuration of devices - peers.
static int	handle_incoming (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock)
	Handle incoming SIP requests (methods).
static int	handle_invite_replaces (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct ast_sockaddr *addr, int *nounlock)
	Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.
static int	handle_request_bye (struct sip_pvt *p, struct sip_request *req)
	Handle incoming BYE request.
static int	handle_request_cancel (struct sip_pvt *p, struct sip_request *req)
	Handle incoming CANCEL request.
static int	handle_request_do (struct sip_request *req, struct ast_sockaddr *addr)
	Handle incoming SIP message - request or response.
static void	handle_request_info (struct sip_pvt *p, struct sip_request *req)
	Receive SIP INFO Message.
static int	handle_request_invite (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct ast_sockaddr *addr, int *recount, const char *e, int *nounlock)
	Handle incoming INVITE request.
static int	handle_request_message (struct sip_pvt *p, struct sip_request *req)
	Handle incoming MESSAGE request.
static int	handle_request_notify (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int seqno, const char *e)
	Handle incoming notifications.
static int	handle_request_options (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.
static int	handle_request_publish (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const int seqno, const char *uri)
static int	handle_request_refer (struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock)
static int	handle_request_register (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Handle incoming REGISTER request.
static int	handle_request_subscribe (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int seqno, const char *e)
	Handle incoming SUBSCRIBE request.
static int	handle_request_update (struct sip_pvt *p, struct sip_request *req)
	bare-bones support for SIP UPDATE
static void	handle_response (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
	Handle SIP response in dialogue.
static void	handle_response_invite (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
	Handle SIP response to INVITE dialogue.
static void	handle_response_notify (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
static void	handle_response_peerpoke (struct sip_pvt *p, int resp, struct sip_request *req)
	Handle qualification responses (OPTIONS).
static void	handle_response_publish (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
static void	handle_response_refer (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
static int	handle_response_register (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
	Handle responses on REGISTER to services.
static void	handle_response_subscribe (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
static void	handle_response_update (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
	Handle authentication challenge for SIP UPDATE.
static int	handle_sip_publish_initial (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const int expires)
static int	handle_sip_publish_modify (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag, const int expires)
static int	handle_sip_publish_refresh (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag, const int expires)
static int	handle_sip_publish_remove (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag)
static int	handle_t38_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v, int *maxdatagram)
	Handle T.38 configuration options common to users and peers.
const char *	hangup_cause2sip (int cause)
	Convert Asterisk hangup causes to SIP codes.
int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes.
static int	init_req (struct sip_request *req, int sipmethod, const char *recip)
	Initialize SIP request.
static int	init_resp (struct sip_request *resp, const char *msg)
	Initialize SIP response, based on SIP request.
static int	initialize_escs (void)
static void	initialize_initreq (struct sip_pvt *p, struct sip_request *req)
	Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.
static void	initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, const char *const explicit_uri)
	Initiate new SIP request to peer/user.
static const char *	insecure2str (int mode)
	Convert Insecure setting to printable string.
static int	interpret_t38_parameters (struct sip_pvt *p, const struct ast_control_t38_parameters *parameters)
	Helper function which updates T.38 capability information and triggers a reinvite.
static int	is_method_allowed (unsigned int *allowed_methods, enum sipmethod method)
	Check if method is allowed for a device or a dialog.
static void	list_route (struct sip_route *route)
	List all routes - mostly for debugging.
static int	load_module (void)
	PBX load module - initialization.
static int	local_attended_transfer (struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno, int *nounlock)
	Find all call legs and bridge transferee with target called from handle_request_refer.
static int	lws2sws (char *msgbuf, int len)
	Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.
static void	make_our_tag (char *tagbuf, size_t len)
	Make our SIP dialog tag.
static int	manager_show_registry (struct mansession *s, const struct message *m)
	Show SIP registrations in the manager API.
static int	manager_sip_qualify_peer (struct mansession *s, const struct message *m)
	Qualify SIP peers in the manager API.
static int	manager_sip_show_peer (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sip_show_peers (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sipnotify (struct mansession *s, const struct message *m)
static int	map_s_x (const struct _map_x_s *table, const char *s, int errorvalue)
	map from a string to an integer value, case insensitive. If no match is found, return errorvalue.
static const char *	map_x_s (const struct _map_x_s *table, int x, const char *errorstring)
	map from an integer value to a string. If no match is found, return errorstring
static void	mark_method_allowed (unsigned int *allowed_methods, enum sipmethod method)
static void	mark_method_unallowed (unsigned int *allowed_methods, enum sipmethod method)
static void	mark_parsed_methods (unsigned int *methods, char *methods_str)
static enum match_req_res	match_req_to_dialog (struct sip_pvt *sip_pvt_ptr, struct match_req_args *arg)
static int	method_match (enum sipmethod id, const char *name)
	returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send
static void	mwi_event_cb (const struct ast_event *event, void *userdata)
	Receive MWI events that we have subscribed to.
static void	network_change_event_cb (const struct ast_event *, void *)
static int	network_change_event_sched_cb (const void *data)
static void	network_change_event_subscribe (void)
static void	network_change_event_unsubscribe (void)
static struct sip_proxy *	obproxy_get (struct sip_pvt *dialog, struct sip_peer *peer)
	Get default outbound proxy or global proxy.
static unsigned int	parse_allowed_methods (struct sip_request *req)
	parse the Allow header to see what methods the endpoint we are communicating with allows.
static void	parse_copy (struct sip_request *dst, const struct sip_request *src)
	Copy SIP request, parse it.
static int	parse_minse (const char *p_hdrval, int *const p_interval)
	Session-Timers: Function for parsing Min-SE header.
static void	parse_moved_contact (struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward)
	Parse 302 Moved temporalily response.
static int	parse_ok_contact (struct sip_pvt *pvt, struct sip_request *req)
	Save contact header for 200 OK on INVITE.
static enum parse_register_result	parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
	Parse contact header and save registration (peer registration).
static int	parse_request (struct sip_request *req)
	Parse a SIP message.
static int	parse_session_expires (const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref)
	Session-Timers: Function for parsing Session-Expires header.
static int	peer_cmp_cb (void *obj, void *arg, int flags)
static int	peer_dump_func (void *userobj, void *arg, int flags)
static int	peer_hash_cb (const void *obj, const int flags)
static int	peer_ipcmp_cb (void *obj, void *arg, int flags)
static int	peer_iphash_cb (const void *obj, const int flags)
static int	peer_is_marked (void *peerobj, void *arg, int flags)
static void	peer_mailboxes_to_str (struct ast_str **mailbox_str, struct sip_peer *peer)
	list peer mailboxes to CLI
static int	peer_markall_func (void *device, void *arg, int flags)
static int	peer_status (struct sip_peer *peer, char *status, int statuslen)
int	peercomparefunc (const void *a, const void *b)
static int	peers_data_provider_get (const struct ast_data_search *search, struct ast_data *data_root)
unsigned int	port_str2int (const char *pt, unsigned int standard)
	converts ascii port to int representation. If no pt buffer is provided or the pt has errors when being converted to an int value, the port provided as the standard is used.
static void	print_codec_to_cli (int fd, struct ast_codec_pref *pref)
	Print codec list from preference to CLI/manager.
static void	print_group (int fd, ast_group_t group, int crlf)
	Print call group and pickup group.
static void	proc_422_rsp (struct sip_pvt *p, struct sip_request *rsp)
	Handle 422 response to INVITE with session-timer requested.
static int	proc_session_timer (const void *vp)
	Session-Timers: Process session refresh timeout event.
static int	process_crypto (struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
static void	process_request_queue (struct sip_pvt *p, int *recount, int *nounlock)
static int	process_sdp (struct sip_pvt *p, struct sip_request *req, int t38action)
	Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().
static int	process_sdp_a_audio (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec)
static int	process_sdp_a_image (const char *a, struct sip_pvt *p)
static int	process_sdp_a_sendonly (const char *a, int *sendonly)
static int	process_sdp_a_text (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
static int	process_sdp_a_video (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)
static int	process_sdp_c (const char *c, struct ast_sockaddr *addr)
static int	process_sdp_o (const char *o, struct sip_pvt *p)
static int	proxy_update (struct sip_proxy *proxy)
static int	publish_expire (const void *data)
static void	pvt_set_needdestroy (struct sip_pvt *pvt, const char *reason)
static int	queue_request (struct sip_pvt *p, const struct sip_request *req)
static struct sip_peer *	realtime_peer (const char *newpeername, struct ast_sockaddr *addr, int devstate_only)
	realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.
static void	realtime_update_peer (const char *peername, struct ast_sockaddr *addr, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms)
	Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.
static void	receive_message (struct sip_pvt *p, struct sip_request *req)
	Receive SIP MESSAGE method messages.
static struct sip_peer *	ref_peer (struct sip_peer *peer, char *tag)
static void	ref_proxy (struct sip_pvt *pvt, struct sip_proxy *proxy)
	maintain proper refcounts for a sip_pvt's outboundproxy
static const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static void	reg_source_db (struct sip_peer *peer)
	Get registration details from Asterisk DB.
static void	register_peer_exten (struct sip_peer *peer, int onoff)
	Automatically add peer extension to dial plan.
static enum check_auth_result	register_verify (struct sip_pvt *p, struct ast_sockaddr *addr, struct sip_request *req, const char *uri)
	Verify registration of user Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth Registration requests are only matched with peers that are marked as "dynamic".
static struct sip_registry *	registry_addref (struct sip_registry *reg, char *tag)
	Add object reference to SIP registry.
static void *	registry_unref (struct sip_registry *reg, char *tag)
static const char *	regstate2str (enum sipregistrystate regstate)
	Convert registration state status to string.
static int	reload (void)
	Part of Asterisk module interface.
static int	reload_config (enum channelreloadreason reason)
	Re-read SIP.conf config file.
static char *	remove_uri_parameters (char *uri)
static int	reply_digest (struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
	reply to authentication for outbound registrations
static int	reqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
	Initialize a SIP request message (not the initial one in a dialog).
static int	resp_needs_contact (const char *msg, enum sipmethod method)
	Test if this response needs a contact header.
static int	respprep (struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Prepare SIP response packet.
static int	restart_monitor (void)
	Start the channel monitor thread.
static void	restart_session_timer (struct sip_pvt *p)
	Session-Timers: Restart session timer.
static int	retrans_pkt (const void *data)
	Retransmit SIP message if no answer (Called from scheduler).
static int	scheduler_process_request_queue (const void *data)
static int	send_provisional_keepalive (const void *data)
static int	send_provisional_keepalive_full (struct sip_pvt *pvt, int with_sdp)
static int	send_provisional_keepalive_with_sdp (const void *data)
static int	send_request (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Send SIP Request to the other part of the dialogue.
static int	send_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
	Transmit response on SIP request.
static enum ast_cc_service_type	service_string_to_service_type (const char *const service_string)
static int	set_address_from_contact (struct sip_pvt *pvt)
	Change the other partys IP address based on given contact.
static void	set_destination (struct sip_pvt *p, char *uri)
	Set destination from SIP URI.
static void	set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
	Parse insecure= setting in sip.conf and set flags according to setting.
static void	set_nonce_randdata (struct sip_pvt *p, int forceupdate)
	builds the sip_pvt's randdata field which is used for the nonce challenge. When forceupdate is not set, the nonce is only updated if the current one is stale. In this case, a stalenonce is one which has already received a response, if a nonce has not received a response it is not always necessary or beneficial to create a new one.
static void	set_peer_defaults (struct sip_peer *peer)
	Set peer defaults before configuring specific configurations.
static unsigned int	set_pvt_allowed_methods (struct sip_pvt *pvt, struct sip_request *req)
static void	set_socket_transport (struct sip_socket *socket, int transport)
static void	set_t38_capabilities (struct sip_pvt *p)
	Set the global T38 capabilities on a SIP dialog structure.
static int	setup_srtp (struct sip_srtp **srtp)
static int	show_channels_cb (void *__cur, void *__arg, int flags)
	callback for show channel|subscription
static int	show_chanstats_cb (void *__cur, void *__arg, int flags)
	Callback for show_chanstats.
static int	sip_addheader (struct ast_channel *chan, const char *data)
	Add a SIP header to an outbound INVITE.
struct sip_pvt *	sip_alloc (ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, struct sip_request *req)
	Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.
static void	sip_alreadygone (struct sip_pvt *dialog)
	Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.
static int	sip_answer (struct ast_channel *ast)
	sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface
static int	sip_call (struct ast_channel *ast, char *dest, int timeout)
	Initiate SIP call from PBX used from the dial() application.
int	sip_cancel_destroy (struct sip_pvt *p)
	Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.
static void	sip_cc_agent_ack (struct ast_cc_agent *agent)
static void	sip_cc_agent_destructor (struct ast_cc_agent *agent)
static int	sip_cc_agent_init (struct ast_cc_agent *agent, struct ast_channel *chan)
static int	sip_cc_agent_recall (struct ast_cc_agent *agent)
static int	sip_cc_agent_start_monitoring (struct ast_cc_agent *agent)
static int	sip_cc_agent_start_offer_timer (struct ast_cc_agent *agent)
static int	sip_cc_agent_status_request (struct ast_cc_agent *agent)
static int	sip_cc_agent_stop_offer_timer (struct ast_cc_agent *agent)
static int	sip_cc_monitor_cancel_available_timer (struct ast_cc_monitor *monitor, int *sched_id)
static void	sip_cc_monitor_destructor (void *private_data)
static int	sip_cc_monitor_request_cc (struct ast_cc_monitor *monitor, int *available_timer_id)
static int	sip_cc_monitor_suspend (struct ast_cc_monitor *monitor)
static int	sip_cc_monitor_unsuspend (struct ast_cc_monitor *monitor)
static char *	sip_cli_notify (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Cli command to send SIP notify to peer.
static int	sip_debug_test_addr (const struct ast_sockaddr *addr)
	See if we pass debug IP filter.
static int	sip_debug_test_pvt (struct sip_pvt *p)
	Test PVT for debugging output.
struct sip_pvt *	sip_destroy (struct sip_pvt *p)
	Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.
static void	sip_destroy_fn (void *p)
static void	sip_destroy_peer (struct sip_peer *peer)
	Destroy peer object from memory.
static void	sip_destroy_peer_fn (void *peer)
static int	sip_devicestate (void *data)
	Part of PBX channel interface.
static char *	sip_do_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Turn on SIP debugging (CLI command).
static char *	sip_do_debug_ip (int fd, const char *arg)
	Enable SIP Debugging for a single IP.
static char *	sip_do_debug_peer (int fd, const char *arg)
	Turn on SIP debugging for a given peer.
static int	sip_do_reload (enum channelreloadreason reason)
	Reload module.
static int	sip_dtmfmode (struct ast_channel *chan, const char *data)
	Set the DTMFmode for an outbound SIP call (application).
static void	sip_dump_history (struct sip_pvt *dialog)
	Dump SIP history to debug log file at end of lifespan for SIP dialog.
static int	sip_epa_register (const struct epa_static_data *static_data)
static int	sip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links
static const char *	sip_get_callid (struct ast_channel *chan)
	Deliver SIP call ID for the call.
static int	sip_get_cc_information (struct sip_request *req, char *subscribe_uri, size_t size, enum ast_cc_service_type *service)
static format_t	sip_get_codec (struct ast_channel *chan)
static enum ast_rtp_glue_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static enum ast_rtp_glue_result	sip_get_trtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
static enum ast_rtp_glue_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static void	sip_handle_cc (struct sip_pvt *pvt, struct sip_request *req, enum ast_cc_service_type service)
static int	sip_hangup (struct ast_channel *ast)
	sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup
static int	sip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.
static int	sip_is_xml_parsable (void)
static int	sip_monitor_instance_cmp_fn (void *obj, void *arg, int flags)
static void	sip_monitor_instance_destructor (void *data)
static int	sip_monitor_instance_hash_fn (const void *obj, const int flags)
static struct sip_monitor_instance *	sip_monitor_instance_init (int core_id, const char *const subscribe_uri, const char *const peername, const char *const device_name)
static const char *	sip_nat_mode (const struct sip_pvt *p)
	Display SIP nat mode.
static struct ast_channel *	sip_new (struct sip_pvt *i, int state, const char *title, const char *linkedid)
	Initiate a call in the SIP channel called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels.
static int	sip_notify_allocate (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_offer_timer_expire (const void *data)
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
	Park a call using the subsystem in res_features.c This is executed in a separate thread.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
static void	sip_peer_hold (struct sip_pvt *p, int hold)
	Change onhold state of a peer using a pvt structure.
static void	sip_poke_all_peers (void)
	Send a poke to all known peers.
static int	sip_poke_noanswer (const void *data)
	React to lack of answer to Qualify poke.
static int	sip_poke_peer (struct sip_peer *peer, int force)
	Check availability of peer, also keep NAT open.
static int	sip_poke_peer_s (const void *data)
	Poke peer (send qualify to check if peer is alive and well).
static int	sip_prepare_socket (struct sip_pvt *p)
static char *	sip_prune_realtime (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Remove temporary realtime objects from memory (CLI).
static char *	sip_qualify_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Send an OPTIONS packet to a SIP peer.
static int	sip_queryoption (struct ast_channel *chan, int option, void *data, int *datalen)
	Query an option on a SIP dialog.
static struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static struct ast_sockaddr *	sip_real_dst (const struct sip_pvt *p)
	The real destination address for a write.
static const char *	sip_reason_code_to_str (enum AST_REDIRECTING_REASON code)
static enum AST_REDIRECTING_REASON	sip_reason_str_to_code (const char *text)
static int	sip_refer_allocate (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_reg_timeout (const void *data)
	Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.
static int	sip_register (const char *value, int lineno)
	create sip_registry object from register=> line in sip.conf and link into reg container
static void	sip_register_tests (void)
	SIP test registration.
static void	sip_registry_destroy (struct sip_registry *reg)
	Destroy registry object Objects created with the register= statement in static configuration.
static int	sip_reinvite_retry (const void *data)
	Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.
static char *	sip_reload (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Force reload of module from cli.
static int	sip_removeheader (struct ast_channel *chan, const char *data)
	Remove SIP headers added previously with SipAddHeader application.
static struct ast_channel *	sip_request_call (const char *type, format_t format, const struct ast_channel *requestor, void *data, int *cause)
	PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.
static int	sip_reregister (const void *data)
	Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
	Read RTP from network.
void	sip_scheddestroy (struct sip_pvt *p, int ms)
	Schedule destruction of SIP dialog.
void	sip_scheddestroy_final (struct sip_pvt *p, int ms)
	Schedule final destruction of SIP dialog. This can not be canceled. This function is used to keep a dialog around for a period of time in order to properly respond to any retransmits.
static void	sip_send_all_mwi_subscriptions (void)
	Send all MWI subscriptions.
static void	sip_send_all_registers (void)
	Send all known registrations.
static int	sip_send_mwi_to_peer (struct sip_peer *peer, const struct ast_event *event, int cache_only)
	Send message waiting indication to alert peer that they've got voicemail.
static int	sip_senddigit_begin (struct ast_channel *ast, char digit)
static int	sip_senddigit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.
static int	sip_sendhtml (struct ast_channel *chan, int subclass, const char *data, int datalen)
	Send message with Access-URL header, if this is an HTML URL only!
static int	sip_sendtext (struct ast_channel *ast, const char *text)
	Send SIP MESSAGE text within a call Called from PBX core sendtext() application.
static char *	sip_set_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Enable/Disable SIP History logging (CLI).
static void	sip_set_redirstr (struct sip_pvt *p, char *reason)
	Translate referring cause.
static int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, format_t codecs, int nat_active)
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
static int	sip_setoption (struct ast_channel *chan, int option, void *data, int datalen)
	Set an option on a SIP dialog.
static char *	sip_show_channel (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show details of one active dialog.
static char *	sip_show_channels (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.
static char *	sip_show_channelstats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	SIP show channelstats CLI (main function).
static char *	sip_show_domains (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI command to list local domains.
static char *	sip_show_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show history details of one dialog.
static char *	sip_show_inuse (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command to show calls within limits set by call_limit.
static char *	sip_show_mwi (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	sip_show_objects (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List all allocated SIP Objects (realtime or static).
static char *	sip_show_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one peer in detail.
static char *	sip_show_peers (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Show Peers command.
static char *	sip_show_registry (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show SIP Registry (registrations with other SIP proxies.
static char *	sip_show_sched (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	sip_show_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List global settings for the SIP channel.
static char *	sip_show_tcp (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show active TCP connections.
static char *	sip_show_user (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one user in detail.
static char *	sip_show_users (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command 'SIP Show Users'.
static int	sip_sipredirect (struct sip_pvt *p, const char *dest)
	Transfer call before connect with a 302 redirect.
static struct sip_st_dlg *	sip_st_alloc (struct sip_pvt *const p)
	Allocate Session-Timers struct w/in dialog.
static int	sip_standard_port (enum sip_transport type, int port)
	Returns the port to use for this socket.
static int	sip_subscribe_mwi (const char *value, int lineno)
	Parse mwi=> line in sip.conf and add to list.
static void	sip_subscribe_mwi_destroy (struct sip_subscription_mwi *mwi)
	Destroy MWI subscription object.
static int	sip_subscribe_mwi_do (const void *data)
	Send a subscription or resubscription for MWI.
static int	sip_t38_abort (const void *data)
	Called to deny a T38 reinvite if the core does not respond to our request.
static struct ast_tcptls_session_instance *	sip_tcp_locate (struct ast_sockaddr *s)
	Find thread for TCP/TLS session (based on IP/Port.
static void *	sip_tcp_worker_fn (void *data)
	SIP TCP connection handler.
static void	sip_tcptls_client_args_destructor (void *obj)
static int	sip_tcptls_write (struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
	used to indicate to a tcptls thread that data is ready to be written
static struct sip_threadinfo *	sip_threadinfo_create (struct ast_tcptls_session_instance *tcptls_session, int transport)
	creates a sip_threadinfo object and links it into the threadt table.
static void	sip_threadinfo_destructor (void *obj)
static int	sip_transfer (struct ast_channel *ast, const char *dest)
	Transfer SIP call.
static char *	sip_unregister (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Unregister (force expiration) a SIP peer in the registry via CLI.
static void	sip_unregister_tests (void)
	SIP test registration.
static int	sip_write (struct ast_channel *ast, struct ast_frame *frame)
	Send frame to media channel (rtp).
static int	sipsock_read (int *id, int fd, short events, void *ignore)
	Read data from SIP UDP socket.
static enum st_mode	st_get_mode (struct sip_pvt *p)
	Get the session-timer mode.
static enum st_refresher	st_get_refresher (struct sip_pvt *p)
	Get the entity (UAC or UAS) that's acting as the session-timer refresher.
static int	st_get_se (struct sip_pvt *p, int max)
	Get Max or Min SE (session timer expiry).
static void	start_session_timer (struct sip_pvt *p)
	Session-Timers: Start session timer.
static void	state_notify_build_xml (int state, int full, const char *exten, const char *context, struct ast_str **tmp, struct sip_pvt *p, int subscribed, const char *mfrom, const char *mto)
	Builds XML portion of NOTIFY messages for presence or dialog updates.
static const char *	stmode2str (enum st_mode m)
static void	stop_media_flows (struct sip_pvt *p)
	Immediately stop RTP, VRTP and UDPTL as applicable.
static void	stop_session_timer (struct sip_pvt *p)
	Session-Timers: Stop session timer.
static int	str2dtmfmode (const char *str)
	maps a string to dtmfmode, returns -1 on error
static enum st_mode	str2stmode (const char *s)
static enum st_refresher	str2strefresher (const char *s)
static const char *	strefresher2str (enum st_refresher r)
static const char *	subscription_type2str (enum subscriptiontype subtype)
	Show subscription type in string format.
static unsigned int	t38_get_rate (enum ast_control_t38_rate rate)
	Get Max T.38 Transmission rate from T38 capabilities.
static void	tcptls_packet_destructor (void *obj)
static struct sip_peer *	temp_peer (const char *name)
	Create temporary peer (used in autocreatepeer mode).
static void	temp_pvt_cleanup (void *)
static int	temp_pvt_init (void *)
static char *	terminate_uri (char *uri)
static int	threadinfo_locate_cb (void *obj, void *arg, int flags)
static int	threadt_cmp_cb (void *obj, void *arg, int flags)
static int	threadt_hash_cb (const void *obj, const int flags)
static char *	transfermode2str (enum transfermodes mode)
	Convert transfer mode to text string.
static int	transmit_cc_notify (struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state)
static void	transmit_fake_auth_response (struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable)
	Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.
static int	transmit_info_with_aoc (struct sip_pvt *p, struct ast_aoc_decoded *decoded)
	Send SIP INFO advice of charge message.
static int	transmit_info_with_digit (struct sip_pvt *p, const char digit, unsigned int duration)
	Send SIP INFO dtmf message, see Cisco documentation on cisco.com.
static int	transmit_info_with_vidupdate (struct sip_pvt *p)
	Send SIP INFO with video update request.
static int	transmit_invite (struct sip_pvt *p, int sipmethod, int sdp, int init, const char *const explicit_uri)
	Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it.
static int	transmit_message_with_text (struct sip_pvt *p, const char *text)
	Transmit text with SIP MESSAGE method.
static int	transmit_notify_with_mwi (struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten)
	Notify user of messages waiting in voicemail (RFC3842).
static int	transmit_notify_with_sipfrag (struct sip_pvt *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer (RFC3515).
static int	transmit_provisional_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp)
static int	transmit_publish (struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char *const explicit_uri)
static int	transmit_refer (struct sip_pvt *p, const char *dest)
	Transmit SIP REFER message (initiated by the transfer() dialplan application.
static int	transmit_register (struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
	Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister()).
static int	transmit_reinvite_with_sdp (struct sip_pvt *p, int t38version, int oldsdp)
	Transmit reinvite with SDP.
static int	transmit_request (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry).
static int	transmit_request_with_auth (struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
	Transmit SIP request, auth added.
static int	transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, no retransmits.
static int	transmit_response_reliable (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.
static int	transmit_response_using_temp (ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
	Transmit response, no retransmits, using a temporary pvt structure.
static int	transmit_response_with_allow (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Append Accept header, content length before transmitting response.
static int	transmit_response_with_auth (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
	Respond with authorization request.
static int	transmit_response_with_date (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Append date and content length before transmitting response.
static int	transmit_response_with_minexpires (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Append Min-Expires header, content length before transmitting response.
static int	transmit_response_with_minse (struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
	Transmit 422 response with Min-SE header (Session-Timers).
static int	transmit_response_with_sdp (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_sip_etag (struct sip_pvt *p, const char *msg, const struct sip_request *req, struct sip_esc_entry *esc_entry, int need_new_etag)
static int	transmit_response_with_t38_sdp (struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_unsupported (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
	Transmit response, no retransmits.
static int	transmit_state_notify (struct sip_pvt *p, int state, int full, int timeout)
	Used in the SUBSCRIBE notification subsystem (RFC3265).
static void	try_suggested_sip_codec (struct sip_pvt *p)
	Try setting codec suggested by the SIP_CODEC channel variable.
static void	unlink_marked_peers_from_tables (void)
static void	unlink_peer_from_tables (struct sip_peer *peer)
static int	unload_module (void)
	PBX unload module API.
static void *	unref_peer (struct sip_peer *peer, char *tag)
static int	update_call_counter (struct sip_pvt *fup, int event)
	update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.
static void	update_connectedline (struct sip_pvt *p, const void *data, size_t datalen)
	Notify peer that the connected line has changed.
static void	update_peer (struct sip_peer *p, int expire)
	Update peer data in database (if used).
static void	update_provisional_keepalive (struct sip_pvt *pvt, int with_sdp)
static void	update_redirecting (struct sip_pvt *p, const void *data, size_t datalen)
	Send a provisional response indicating that a call was redirected.
Defines
#define	SIP_PEDANTIC_DECODE(str)
Variables
static int	can_parse_xml
static unsigned int	chan_idx
static const char	config [] = "sip.conf"
static int	default_expiry = DEFAULT_DEFAULT_EXPIRY
static struct ast_jb_conf	default_jbconf
	Global jitterbuffer configuration - by default, jb is disabled.
static unsigned int	dumphistory
static int	global_authfailureevents
static unsigned int	global_autoframing
static int	global_callcounter
static unsigned int	global_cos_audio
static unsigned int	global_cos_sip
static unsigned int	global_cos_text
static unsigned int	global_cos_video
static int	global_dynamic_exclude_static = 0
static struct ast_jb_conf	global_jbconf
static int	global_match_auth_username
static int	global_max_se
static int	global_min_se
static int	global_prematuremediafilter
static int	global_qualify_gap
static int	global_qualify_peers
static int	global_qualifyfreq
static int	global_reg_timeout
static int	global_regattempts_max
static int	global_relaxdtmf
static int	global_rtpholdtimeout
static int	global_rtpkeepalive
static int	global_rtptimeout
static char	global_sdpowner [AST_MAX_EXTENSION]
static char	global_sdpsession [AST_MAX_EXTENSION]
static int	global_shrinkcallerid
static enum st_mode	global_st_mode
static enum st_refresher	global_st_refresher
static int	global_t1
static int	global_t1min
static int	global_timer_b
static unsigned int	global_tos_audio
static unsigned int	global_tos_sip
static unsigned int	global_tos_text
static unsigned int	global_tos_video
static char	global_useragent [AST_MAX_EXTENSION]
static struct invstate2stringtable	invitestate2string []
	Readable descriptions of device states.
static int	max_expiry = DEFAULT_MAX_EXPIRY
static int	min_expiry = DEFAULT_MIN_EXPIRY
static int	mwi_expiry = DEFAULT_MWI_EXPIRY
static const char	notify_config [] = "sip_notify.conf"
static unsigned int	recordhistory
static struct sip_settings	sip_cfg
static struct cfsip_methods	sip_methods []
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type.
static struct sip_reasons	sip_reason_table []
	Diversion header reasons.
static struct cfsubscription_types	subscription_types []
	Subscription types that we support. We support dialoginfo updates (really device status, not dialog info as was the original intent of the standard) SIMPLE presence used for device status Voicemail notification subscriptions.
DefaultSettings
Default setttings are used as a channel setting and as a default when configuring devices
static char	default_callerid [AST_MAX_EXTENSION]
static char	default_engine [256]
static char	default_fromdomain [AST_MAX_EXTENSION]
static int	default_fromdomainport
static char	default_language [MAX_LANGUAGE]
static int	default_maxcallbitrate
static char	default_mohinterpret [MAX_MUSICCLASS]
static char	default_mohsuggest [MAX_MUSICCLASS]
static char	default_mwi_from [80]
static char	default_notifymime [AST_MAX_EXTENSION]
static char	default_parkinglot [AST_MAX_CONTEXT]
static struct ast_codec_pref	default_prefs
static unsigned int	default_primary_transport
static int	default_qualify
static unsigned int	default_transports
static char	default_vmexten [AST_MAX_EXTENSION]

---

Detailed Description

Implementation of Session Initiation Protocol.

Author:

Mark Spencer <markster@digium.com>

See Also:

• CREDITS

Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support Configuration file sip.conf

********** IMPORTANT *

Note:

TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration settings, dialplan commands and dialplans apps/functions See SIP TCP and TLS support

******** General TODO:s

Todo:

Better support of forking

Todo:

VIA branch tag transaction checking

Todo:

Transaction support

******** Wishlist: Improvements

• Support of SIP domains for devices, so that we match on username in the From: header
• Connect registrations with a specific device on the incoming call. It's not done automatically in Asterisk

Overview of the handling of SIP sessions

The SIP channel handles several types of SIP sessions, or dialogs, not all of them being "telephone calls".

• Incoming calls that will be sent to the PBX core
• Outgoing calls, generated by the PBX
• SIP subscriptions and notifications of states and voicemail messages
• SIP registrations, both inbound and outbound
• SIP peer management (peerpoke, OPTIONS)
• SIP text messages

In the SIP channel, there's a list of active SIP dialogs, which includes all of these when they are active. "sip show channels" in the CLI will show most of these, excluding subscriptions which are shown by "sip show subscriptions"

incoming packets

Incoming packets are received in the monitoring thread, then handled by sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread. sipsock_read() function parses the packet and matches an existing dialog or starts a new SIP dialog.

sipsock_read sends the packet to handle_incoming(), that parses a bit more. If it is a response to an outbound request, the packet is sent to handle_response(). If it is a request, handle_incoming() sends it to one of a list of functions depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc sipsock_read locks the ast_channel if it exists (an active call) and unlocks it after we have processed the SIP message.

A new INVITE is sent to handle_request_invite(), that will end up starting a new channel in the PBX, the new channel after that executing in a separate channel thread. This is an incoming "call". When the call is answered, either by a bridged channel or the PBX itself the sip_answer() function is called.

The actual media - Video or Audio - is mostly handled by the RTP subsystem in rtp.c

Outbound calls

Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up

The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

---

Define Documentation

#define append_history	(	p,
		event,
		fmt,
		args...		)	append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history.

Returns:

Always returns 0

Definition at line 2170 of file chan_sip.c.

Referenced by __sip_autodestruct(), __sip_reliable_xmit(), auto_congest(), build_reply_digest(), cb_extensionstate(), check_auth(), do_register_auth(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response(), handle_response_invite(), local_attended_transfer(), obproxy_get(), process_sdp(), pvt_set_needdestroy(), retrans_pkt(), send_request(), send_response(), sip_cancel_destroy(), sip_fixup(), sip_hangup(), sip_new(), sip_park_thread(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), transmit_register(), transmit_reinvite_with_sdp(), transmit_response_with_auth(), and update_connectedline().

#define CHECK_AUTH_BUF_INITLEN   256

Definition at line 13245 of file chan_sip.c.

Referenced by check_auth(), and transmit_fake_auth_response().

#define check_request_transport	(	peer,
		tmpl		)

generic function for determining if a correct transport is being used to contact a peer

this is done as a macro so that the "tmpl" var can be passed either a sip_request or a sip_peer

Definition at line 2253 of file chan_sip.c.

Referenced by create_addr_from_peer(), and register_verify().

#define DATA_EXPORT_SIP_PEER

(

MEMBER

)

Definition at line 28207 of file chan_sip.c.

#define FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-39.39s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-25.25s %-39.39s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-47.47s %-9.9s %-6.6s\n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 17056 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-10.10s %-10.10s\n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT2   "%-39.39s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-10s %s\n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT2   "%-47.47s %9.9s %6.6s\n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 17055 of file chan_sip.c.

#define FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"

Definition at line 17054 of file chan_sip.c.

Referenced by sip_show_channels().

#define FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"

Definition at line 17053 of file chan_sip.c.

Referenced by show_channels_cb().

#define SIP_PEDANTIC_DECODE

(

str

)

Value:

if (sip_cfg.pedanticsipchecking && !ast_strlen_zero(str)) { \
      ast_uri_decode(str); \
   }  \

Definition at line 696 of file chan_sip.c.

Referenced by check_user_full(), get_also_info(), get_destination(), get_refer_info(), and register_verify().

#define sip_pvt_lock

(

x

)

ao2_lock(x)

Definition at line 1077 of file chan_sip.c.

Referenced by __sip_ack(), add_header_max_forwards(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), find_call(), get_sip_pvt_byid_locked(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_request_subscribe(), local_attended_transfer(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_call(), sip_cc_agent_ack(), sip_cc_agent_destructor(), sip_cc_agent_init(), sip_cc_agent_recall(), sip_cc_monitor_request_cc(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_monitor_instance_destructor(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), transmit_publish(), and update_call_counter().

#define sip_pvt_trylock

(

x

)

ao2_trylock(x)

Definition at line 1078 of file chan_sip.c.

Referenced by dialog_needdestroy(), and sip_hangup().

#define sip_pvt_unlock

(

x

)

ao2_unlock(x)

Definition at line 1079 of file chan_sip.c.

Referenced by __sip_ack(), add_header_max_forwards(), auto_congest(), cb_extensionstate(), check_rtp_timeout(), complete_sipch(), dialog_needdestroy(), get_sip_pvt_byid_locked(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_subscribe(), local_attended_transfer(), proc_session_timer(), retrans_pkt(), scheduler_process_request_queue(), sip_answer(), sip_call(), sip_cc_agent_ack(), sip_cc_agent_destructor(), sip_cc_agent_init(), sip_cc_agent_recall(), sip_cc_monitor_request_cc(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_monitor_instance_destructor(), sip_new(), sip_park(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_senddigit_begin(), sip_senddigit_end(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), transmit_publish(), and update_call_counter().

#define UNLINK	(	element,
		head,
		prev		)

some list management macros.

Definition at line 1176 of file chan_sip.c.

Referenced by __sip_ack(), handle_request_cancel(), and retrans_pkt().

---

Enumeration Type Documentation

enum match_req_res

Enumerator:

SIP_REQ_MATCH
SIP_REQ_NOT_MATCH
SIP_REQ_LOOP_DETECTED

Definition at line 7193 of file chan_sip.c.

                   {
   SIP_REQ_MATCH,
   SIP_REQ_NOT_MATCH,
   SIP_REQ_LOOP_DETECTED,
};

---

Function Documentation

static const char * __get_header	(	const struct sip_request *	req,
		const char *	name,
		int *	start
	)			[static]

Definition at line 6778 of file chan_sip.c.

References ast_skip_blanks(), find_alias(), len(), pass, and sip_cfg.

{
   int pass;

   /*
    * Technically you can place arbitrary whitespace both before and after the ':' in
    * a header, although RFC3261 clearly says you shouldn't before, and place just
    * one afterwards.  If you shouldn't do it, what absolute idiot decided it was
    * a good idea to say you can do it, and if you can do it, why in the hell would.
    * you say you shouldn't.
    * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
    * and we always allow spaces after that for compatibility.
    */
   for (pass = 0; name && pass < 2;pass++) {
      int x, len = strlen(name);
      for (x = *start; x < req->headers; x++) {
         const char *header = REQ_OFFSET_TO_STR(req, header[x]);
         if (!strncasecmp(header, name, len)) {
            const char *r = header + len; /* skip name */
            if (sip_cfg.pedanticsipchecking)
               r = ast_skip_blanks(r);

            if (*r == ':') {
               *start = x+1;
               return ast_skip_blanks(r+1);
            }
         }
      }
      if (pass == 0) /* Try aliases */
         name = find_alias(name, NULL);
   }

   /* Don't return NULL, so get_header is always a valid pointer */
   return "";
}

static void __init_check_auth_buf

(

void

)

[static]

Definition at line 13244 of file chan_sip.c.

{

static void __init_ts_temp_pvt

(

void

)

[static]

A per-thread temporary pvt structure.

Definition at line 1102 of file chan_sip.c.

: The address we bind to */

static void __reg_module

(

void

)

[static]

Definition at line 28645 of file chan_sip.c.

static int __set_address_from_contact	(	const char *	fullcontact,
		struct ast_sockaddr *	addr,
		int	tcp
	)			[static]

Definition at line 12835 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, get_transport_str2enum(), and LOG_WARNING.

Referenced by build_peer(), and set_address_from_contact().

{
   char *domain, *transport;
   char contact_buf[256];
   char *contact;

   /* Work on a copy */
   ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
   contact = contact_buf;

   /* 
    * We have only the part in <brackets> here so we just need to parse a SIP URI.
    *
    * Note: The outbound proxy could be using UDP between the proxy and Asterisk.
    * We still need to be able to send to the remote agent through the proxy.
    */

   if (parse_uri(contact, "sip:,sips:", &contact, NULL, &domain,
            &transport)) {
      ast_log(LOG_WARNING, "Invalid contact uri %s (missing sip: or sips:), attempting to use anyway\n", fullcontact);
   }

   /* XXX This could block for a long time XXX */
   /* We should only do this if it's a name, not an IP */
   /* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
      to find transport, port address and hostname. If there's a port number, we have to
      assume that the domain part is a host name and only look for an A/AAAA record in DNS.
   */

   if (ast_sockaddr_resolve_first(addr, domain, 0)) {
      ast_log(LOG_WARNING, "Invalid host name in Contact: (can't "
         "resolve in DNS) : '%s'\n", domain);
      return -1;
   }

   /* set port */
   if (!ast_sockaddr_port(addr)) {
      ast_sockaddr_set_port(addr,
                  (get_transport_str2enum(transport) ==
                   SIP_TRANSPORT_TLS ||
                   !strncasecmp(fullcontact, "sips", 4)) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }

   return 0;
}

int __sip_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)

Acknowledges receipt of a packet and stops retransmission called with p locked.

Definition at line 3700 of file chan_sip.c.

References ast_debug, ast_free, ast_sched_del(), FALSE, ref_proxy(), sip_pvt_lock, sip_pvt_unlock, TRUE, and UNLINK.

Referenced by __sip_pretend_ack(), handle_incoming(), handle_request_invite(), handle_request_publish(), and handle_response().

{
   struct sip_pkt *cur, *prev = NULL;
   const char *msg = "Not Found";   /* used only for debugging */
   int res = FALSE;

   /* If we have an outbound proxy for this dialog, then delete it now since
     the rest of the requests in this dialog needs to follow the routing.
     If obforcing is set, we will keep the outbound proxy during the whole
     dialog, regardless of what the SIP rfc says
   */
   if (p->outboundproxy && !p->outboundproxy->force){
      ref_proxy(p, NULL);
   }

   for (cur = p->packets; cur; prev = cur, cur = cur->next) {
      if (cur->seqno != seqno || cur->is_resp != resp) {
         continue;
      }
      if (cur->is_resp || cur->method == sipmethod) {
         res = TRUE;
         msg = "Found";
         if (!resp && (seqno == p->pendinginvite)) {
            ast_debug(1, "Acked pending invite %d\n", p->pendinginvite);
            p->pendinginvite = 0;
         }
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
         }
         /* This odd section is designed to thwart a
          * race condition in the packet scheduler. There are
          * two conditions under which deleting the packet from the
          * scheduler can fail.
          *
          * 1. The packet has been removed from the scheduler because retransmission
          * is being attempted. The problem is that if the packet is currently attempting
          * retransmission and we are at this point in the code, then that MUST mean
          * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
          * lock temporarily to allow retransmission.
          *
          * 2. The packet has reached its maximum number of retransmissions and has
          * been permanently removed from the packet scheduler. If this is the case, then
          * the packet's retransid will be set to -1. The atomicity of the setting and checking
          * of the retransid to -1 is ensured since in both cases p's lock is held.
          */
         while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         }
         UNLINK(cur, p->packets, prev);
         dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
         if (cur->data) {
            ast_free(cur->data);
         }
         ast_free(cur);
         break;
      }
   }
   ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n",
      p->callid, resp ? "Response" : "Request", seqno, msg);
   return res;
}

static int __sip_autodestruct

(

const void *

data

)

[static]

Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().

Definition at line 3572 of file chan_sip.c.

References __sip_pretend_ack(), append_history, AST_CAUSE_PROTOCOL_ERROR, ast_debug, AST_EXTENSION_DEACTIVATED, ast_log(), ast_queue_hangup_with_cause(), dialog_unlink_all(), LOG_WARNING, method_match(), NONE, pvt_set_needdestroy(), sip_methods, sip_scheddestroy(), cfsip_methods::text, transmit_request_with_auth(), transmit_state_notify(), and TRUE.

Referenced by sip_scheddestroy(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *)data;

   /* If this is a subscription, tell the phone that we got a timeout */
   if (p->subscribed && p->subscribed != MWI_NOTIFICATION && p->subscribed != CALL_COMPLETION) {
      transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE);  /* Send last notification */
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "timeout");
      ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
      return 10000;  /* Reschedule this destruction so that we know that it's gone */
   }

   /* If there are packets still waiting for delivery, delay the destruction */
   if (p->packets) {
      if (!p->needdestroy) {
         char method_str[31];
         ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
         append_history(p, "ReliableXmit", "timeout");
         if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
            if (method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
               pvt_set_needdestroy(p, "autodestruct");
            }
         }
         return 10000;
      } else {
         /* They've had their chance to respond. Time to bail */
         __sip_pretend_ack(p);
      }
   }

   /* Reset schedule ID */
   p->autokillid = -1;

   if (p->owner) {
      ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
      ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
   } else if (p->refer && !p->alreadygone) {
      ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
      transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
      append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      append_history(p, "AutoDestroy", "%s", p->callid);
      ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
      dialog_unlink_all(p, TRUE, TRUE); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
      /* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
      /* sip_destroy(p); */      /* Go ahead and destroy dialog. All attempts to recover is done */
      /* sip_destroy also absorbs the reference */
   }
   dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
   return 0;
}

void __sip_destroy	(	struct sip_pvt *	p,
		int	lockowner,
		int	lockdialoglist
	)

Execute destruction of SIP dialog structure, release memory.

Definition at line 5249 of file chan_sip.c.

References ao2_ref, ast_cc_config_params_destroy(), ast_channel_lock, ast_channel_unlock, ast_debug, ast_free, ast_free_ha(), AST_LIST_REMOVE_HEAD, ast_rtp_instance_destroy(), AST_SCHED_DEL_UNREF, AST_SOFTHANGUP_DEV, ast_string_field_free_memory, ast_test_flag, ast_udptl_destroy(), ast_variables_destroy(), ast_verbose, deinit_req(), free_old_route(), registry_unref(), sip_debug_test_pvt(), sip_dump_history(), sip_methods, cfsip_methods::text, TRUE, unref_peer(), and update_call_counter().

Referenced by sip_destroy().

{
   struct sip_request *req;

   if (p->stimer) {
      ast_free(p->stimer);
      p->stimer = NULL;
   }

   if (sip_debug_test_pvt(p))
      ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      update_call_counter(p, DEC_CALL_LIMIT);
      ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
   }

   /* Unlink us from the owner if we have one */
   if (p->owner) {
      if (lockowner)
         ast_channel_lock(p->owner);
      ast_debug(1, "Detaching from %s\n", p->owner->name);
      p->owner->tech_pvt = NULL;
      /* Make sure that the channel knows its backend is going away */
      p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
      if (lockowner)
         ast_channel_unlock(p->owner);
      /* Give the channel a chance to react before deallocation */
      usleep(1);
   }

   /* Remove link from peer to subscription of MWI */
   if (p->relatedpeer && p->relatedpeer->mwipvt)
      p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   if (p->relatedpeer && p->relatedpeer->call == p)
      p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   
   if (p->relatedpeer)
      p->relatedpeer = unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
   
   if (p->registry) {
      if (p->registry->call == p)
         p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
      p->registry = registry_unref(p->registry, "delete p->registry");
   }
   
   if (p->mwi) {
      p->mwi->call = NULL;
   }

   if (dumphistory)
      sip_dump_history(p);

   if (p->options)
      ast_free(p->options);

   if (p->notify) {
      ast_variables_destroy(p->notify->headers);
      ast_free(p->notify->content);
      ast_free(p->notify);
   }
   if (p->rtp) {
      ast_rtp_instance_destroy(p->rtp);
   }
   if (p->vrtp) {
      ast_rtp_instance_destroy(p->vrtp);
   }
   if (p->trtp) {
      ast_rtp_instance_destroy(p->trtp);
   }
   if (p->udptl)
      ast_udptl_destroy(p->udptl);
   if (p->refer)
      ast_free(p->refer);
   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }
   deinit_req(&p->initreq);

   /* Destroy Session-Timers if allocated */
   if (p->stimer) {
      p->stimer->quit_flag = 1;
      if (p->stimer->st_active == TRUE && p->stimer->st_schedid > -1) {
         AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
               dialog_unref(p, "removing session timer ref"));
      }
      ast_free(p->stimer);
      p->stimer = NULL;
   }

   /* Clear history */
   if (p->history) {
      struct sip_history *hist;
      while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
         ast_free(hist);
         p->history_entries--;
      }
      ast_free(p->history);
      p->history = NULL;
   }

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      ast_free(req);
   }

   if (p->chanvars) {
      ast_variables_destroy(p->chanvars);
      p->chanvars = NULL;
   }

   if (p->srtp) {
      sip_srtp_destroy(p->srtp);
      p->srtp = NULL;
   }

   if (p->vsrtp) {
      sip_srtp_destroy(p->vsrtp);
      p->vsrtp = NULL;
   }

   if (p->tsrtp) {
      sip_srtp_destroy(p->tsrtp);
      p->tsrtp = NULL;
   }

   if (p->directmediaha) {
      ast_free_ha(p->directmediaha);
      p->directmediaha = NULL;
   }

   ast_string_field_free_memory(p);

   ast_cc_config_params_destroy(p->cc_params);

   if (p->epa_entry) {
      ao2_ref(p->epa_entry, -1);
      p->epa_entry = NULL;
   }

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   }
}

static int __sip_do_register

(

struct sip_registry *

r

)

[static]

Register with SIP proxy.

Returns:

see __sip_xmit

Definition at line 12111 of file chan_sip.c.

References transmit_register().

Referenced by sip_reregister().

{
   int res;

   res = transmit_register(r, SIP_REGISTER, NULL, NULL);
   return res;
}

void __sip_pretend_ack

(

struct sip_pvt *

p

)

Pretend to ack all packets called with p locked.

Definition at line 3767 of file chan_sip.c.

References __sip_ack(), ast_log(), find_sip_method(), LOG_WARNING, sip_methods, and cfsip_methods::text.

Referenced by __sip_autodestruct(), handle_request_bye(), handle_request_cancel(), and sip_reg_timeout().

{
   struct sip_pkt *cur = NULL;

   while (p->packets) {
      int method;
      if (cur == p->packets) {
         ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
         return;
      }
      cur = p->packets;
      method = (cur->method) ? cur->method : find_sip_method(cur->data->str);
      __sip_ack(p, cur->seqno, cur->is_resp, method);
   }
}

static enum sip_result __sip_reliable_xmit	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		struct ast_str *	data,
		int	len,
		int	fatal,
		int	sipmethod
	)			[static]

Transmit packet with retransmits.

Returns:

0 on success, -1 on failure to allocate packet

Todo:

According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited

Definition at line 3480 of file chan_sip.c.

References __sip_xmit(), append_history, ast_calloc, ast_debug, ast_free, ast_log(), AST_PTHREADT_NULL, AST_SCHED_DEL, AST_SCHED_REPLACE_VARIABLE, ast_str_buffer(), ast_str_create(), ast_str_set(), ast_tvnow(), DEFAULT_RETRANS, LOG_ERROR, and retrans_pkt().

Referenced by send_request(), and send_response().

{
   struct sip_pkt *pkt = NULL;
   int siptimer_a = DEFAULT_RETRANS;
   int xmitres = 0;
   int respid;

   if (sipmethod == SIP_INVITE) {
      /* Note this is a pending invite */
      p->pendinginvite = seqno;
   }

   /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
   /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
   /*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
   if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
      xmitres = __sip_xmit(p, data, len); /* Send packet */
      if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
         append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
         return AST_FAILURE;
      } else {
         return AST_SUCCESS;
      }
   }

   if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1))) {
      return AST_FAILURE;
   }
   /* copy data, add a terminator and save length */
   if (!(pkt->data = ast_str_create(len))) {
      ast_free(pkt);
      return AST_FAILURE;
   }
   ast_str_set(&pkt->data, 0, "%s%s", data->str, "\0");
   pkt->packetlen = len;
   /* copy other parameters from the caller */
   pkt->method = sipmethod;
   pkt->seqno = seqno;
   pkt->is_resp = resp;
   pkt->is_fatal = fatal;
   pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
   pkt->next = p->packets;
   p->packets = pkt; /* Add it to the queue */
   if (resp) {
      /* Parse out the response code */
      if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
         pkt->response_code = respid;
      }
   }
   pkt->timer_t1 = p->timer_t1;  /* Set SIP timer T1 */
   pkt->retransid = -1;
   if (pkt->timer_t1) {
      siptimer_a = pkt->timer_t1;
   }

   pkt->time_sent = ast_tvnow(); /* time packet was sent */
   pkt->retrans_stop_time = 64 * (pkt->timer_t1 ? pkt->timer_t1 : DEFAULT_TIMER_T1); /* time in ms after pkt->time_sent to stop retransmission */

   /* Schedule retransmission */
   AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
   if (sipdebug) {
      ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);
   }

   xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);   /* Send packet */

   if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
      ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
      AST_SCHED_DEL(sched, pkt->retransid);
      p->packets = pkt->next;
      pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
      ast_free(pkt->data);
      ast_free(pkt);
      return AST_FAILURE;
   } else {
      /* This is odd, but since the retrans timer starts at 500ms and the do_monitor thread
       * only wakes up every 1000ms by default, we have to poke the thread here to make
       * sure it successfully detects this must be retransmitted in less time than
       * it usually sleeps for. Otherwise it might not retransmit this packet for 1000ms. */
      if (monitor_thread != AST_PTHREADT_NULL) {
         pthread_kill(monitor_thread, SIGURG);
      }
      return AST_SUCCESS;
   }
}

int __sip_semi_ack	(	struct sip_pvt *	p,
		int	seqno,
		int	resp,
		int	sipmethod
	)

Acks receipt of packet, keep it around (used for provisional responses).

Definition at line 3784 of file chan_sip.c.

References ast_debug, AST_SCHED_DEL, FALSE, method_match(), sip_methods, cfsip_methods::text, and TRUE.

Referenced by handle_response(), and sip_hangup().

{
   struct sip_pkt *cur;
   int res = FALSE;

   for (cur = p->packets; cur; cur = cur->next) {
      if (cur->seqno == seqno && cur->is_resp == resp &&
         (cur->is_resp || method_match(sipmethod, cur->data->str))) {
         /* this is our baby */
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
         }
         AST_SCHED_DEL(sched, cur->retransid);
         res = TRUE;
         break;
      }
   }
   ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
   return res;
}

static int __sip_subscribe_mwi_do

(

struct sip_subscription_mwi *

mwi

)

[static]

Actually setup an MWI subscription or resubscribe.

Definition at line 11460 of file chan_sip.c.

References ast_dnsmgr_lookup(), ast_set_flag, ast_sip_ouraddrfor(), ast_sockaddr_port, ast_sockaddr_set_port, ast_string_field_set, ast_strlen_zero(), ASTOBJ_REF, bindaddr, build_callid_pvt(), build_contact(), build_via(), create_addr(), dialog_unlink_all(), get_address_family_filter(), get_srv_protocol(), get_srv_service(), MAXHOSTNAMELEN, obproxy_get(), ref_proxy(), set_socket_transport(), sip_alloc(), sip_cfg, transmit_invite(), and TRUE.

Referenced by sip_subscribe_mwi_do().

{
   /* If we have no DNS manager let's do a lookup */
   if (!mwi->dnsmgr) {
      char transport[MAXHOSTNAMELEN];
      snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(mwi->transport), get_srv_protocol(mwi->transport));

      mwi->us.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
      ast_dnsmgr_lookup(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
   }

   /* If we already have a subscription up simply send a resubscription */
   if (mwi->call) {
      transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 0, NULL);
      return 0;
   }
   
   /* Create a dialog that we will use for the subscription */
   if (!(mwi->call = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL))) {
      return -1;
   }

   ref_proxy(mwi->call, obproxy_get(mwi->call, NULL));

   if (!ast_sockaddr_port(&mwi->us) && mwi->portno) {
      ast_sockaddr_set_port(&mwi->us, mwi->portno);
   }
   
   /* Setup the destination of our subscription */
   if (create_addr(mwi->call, mwi->hostname, &mwi->us, 0, NULL)) {
      dialog_unlink_all(mwi->call, TRUE, TRUE);
      mwi->call = dialog_unref(mwi->call, "unref dialog after unlink_all");
      return 0;
   }

   mwi->call->expiry = mwi_expiry;
   
   if (!mwi->dnsmgr && mwi->portno) {
      ast_sockaddr_set_port(&mwi->call->sa, mwi->portno);
      ast_sockaddr_set_port(&mwi->call->recv, mwi->portno);
   } else {
      mwi->portno = ast_sockaddr_port(&mwi->call->sa);
   }
   
   /* Set various other information */
   if (!ast_strlen_zero(mwi->authuser)) {
      ast_string_field_set(mwi->call, peername, mwi->authuser);
      ast_string_field_set(mwi->call, authname, mwi->authuser);
      ast_string_field_set(mwi->call, fromuser, mwi->authuser);
   } else {
      ast_string_field_set(mwi->call, peername, mwi->username);
      ast_string_field_set(mwi->call, authname, mwi->username);
      ast_string_field_set(mwi->call, fromuser, mwi->username);
   }
   ast_string_field_set(mwi->call, username, mwi->username);
   if (!ast_strlen_zero(mwi->secret)) {
      ast_string_field_set(mwi->call, peersecret, mwi->secret);
   }
   set_socket_transport(&mwi->call->socket, mwi->transport);
   mwi->call->socket.port = htons(mwi->portno);
   ast_sip_ouraddrfor(&mwi->call->sa, &mwi->call->ourip, mwi->call);
   build_contact(mwi->call);
   build_via(mwi->call);
   build_callid_pvt(mwi->call);
   ast_set_flag(&mwi->call->flags[0], SIP_OUTGOING);
   
   /* Associate the call with us */
   mwi->call->mwi = ASTOBJ_REF(mwi);
   
   /* Actually send the packet */
   transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 2, NULL);

   return 0;
}

static int __sip_xmit	(	struct sip_pvt *	p,
		struct ast_str *	data,
		int	len
	)			[static]

Transmit SIP message Sends a SIP request or response on a given socket (in the pvt) Called by retrans_pkt, send_request, send_response and __sip_reliable_xmit.

Returns:

length of transmitted message, XMIT_ERROR on known network failures -1 on other failures.

Definition at line 3081 of file chan_sip.c.

References ast_debug, ast_log(), ast_sendto(), ast_sockaddr_stringify(), errno, get_transport_pvt(), LOG_WARNING, sip_prepare_socket(), sip_real_dst(), and sip_tcptls_write().

Referenced by __sip_reliable_xmit(), retrans_pkt(), send_request(), and send_response().

{
   int res = 0;
   const struct ast_sockaddr *dst = sip_real_dst(p);

   ast_debug(2, "Trying to put '%.11s' onto %s socket destined for %s\n", data->str, get_transport_pvt(p), ast_sockaddr_stringify(dst));

   if (sip_prepare_socket(p) < 0) {
      return XMIT_ERROR;
   }

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      res = ast_sendto(p->socket.fd, data->str, len, 0, dst);
   } else if (p->socket.tcptls_session) {
      res = sip_tcptls_write(p->socket.tcptls_session, data->str, len);
   } else {
      ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
      return XMIT_ERROR;
   }

   if (res == -1) {
      switch (errno) {
      case EBADF:       /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
      case EHOSTUNREACH:   /* Host can't be reached */
      case ENETDOWN:       /* Interface down */
      case ENETUNREACH: /* Network failure */
      case ECONNREFUSED:      /* ICMP port unreachable */
         res = XMIT_ERROR; /* Don't bother with trying to transmit again */
      }
   }
   if (res != len) {
      ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s returned %d: %s\n", data, len, ast_sockaddr_stringify(dst), res, strerror(errno));
   }

   return res;
}

static int __transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)			[static]

Base transmit response function.

Definition at line 9642 of file chan_sip.c.

References add_cc_call_info_to_response(), add_header(), add_rpid(), ast_cause2str(), ast_clear_flag, ast_log(), ast_test_flag, get_header(), hangup_sip2cause(), LOG_WARNING, respprep(), and send_response().

Referenced by transmit_response(), transmit_response_reliable(), and transmit_response_using_temp().

{
   struct sip_request resp;
   int seqno = 0;

   if (reliable && (sscanf(get_header(req, "CSeq"), "%30d ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);

   if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
         && ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
         && (!strncmp(msg, "180", 3) || !strncmp(msg, "183", 3))) {
      ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
      add_rpid(&resp, p);
   }
   if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
      add_cc_call_info_to_response(p, &resp);
   }

   /* If we are cancelling an incoming invite for some reason, add information
      about the reason why we are doing this in clear text */
   if (p->method == SIP_INVITE && msg[0] != '1') {
      char buf[20];

      if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON)) {
         int hangupcause = 0;

         if (p->owner && p->owner->hangupcause) {
            hangupcause = p->owner->hangupcause;
         } else if (p->hangupcause) {
            hangupcause = p->hangupcause;
         } else {
            int respcode;
            if (sscanf(msg, "%30d ", &respcode))
               hangupcause = hangup_sip2cause(respcode);
         }

         if (hangupcause) {
            sprintf(buf, "Q.850;cause=%i", hangupcause & 0x7f);
            add_header(&resp, "Reason", buf);
         }
      }

      if (p->owner && p->owner->hangupcause) {
         add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
         snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
         add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
      }
   }
   return send_response(p, &resp, reliable, seqno);
}

static void __unreg_module

(

void

)

[static]

Definition at line 28645 of file chan_sip.c.

static char * _sip_qualify_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Send qualify message to peer from cli or manager. Mostly for debugging.

Definition at line 16112 of file chan_sip.c.

References ast_cli(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, find_peer(), sip_poke_peer(), TRUE, and unref_peer().

Referenced by manager_sip_qualify_peer(), and sip_qualify_peer().

{
   struct sip_peer *peer;
   int load_realtime;

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   if ((peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0))) {
      sip_poke_peer(peer, 1);
      unref_peer(peer, "qualify: done with peer");
   } else if (type == 0) {
      ast_cli(fd, "Peer '%s' not found\n", argv[3]);
   } else {
      astman_send_error(s, m, "Peer not found");
   }
   return CLI_SUCCESS;
}

static char * _sip_show_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Show one peer in detail (main function).

Definition at line 16197 of file chan_sip.c.

References ARRAY_LEN, ast_callerid_merge(), ast_cdr_flags2str(), ast_check_realtime(), ast_cli(), AST_CLI_YESNO, ast_codec_pref_index(), ast_describe_caller_presentation(), ast_getformatname(), ast_getformatname_multiple(), ast_print_group(), ast_sched_when(), ast_sockaddr_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_str_alloca, ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, dtmfmode2str(), FALSE, faxec2str(), find_peer(), get_transport(), get_transport_list(), insecure2str(), ast_variable::name, ast_variable::next, peer_mailboxes_to_str(), peer_status(), print_codec_to_cli(), print_group(), S_OR, sip_cfg, status, stmode2str(), strefresher2str(), text, transfermode2str(), TRUE, unref_peer(), and ast_variable::value.

Referenced by manager_sip_show_peer(), and sip_show_peer().

{
   char status[30] = "";
   char cbuf[256];
   struct sip_peer *peer;
   char codec_buf[512];
   struct ast_codec_pref *pref;
   struct ast_variable *v;
   struct sip_auth *auth;
   int x = 0, load_realtime;
   format_t codec = 0;
   int realtimepeers;

   realtimepeers = ast_check_realtime("sippeers");

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0);

   if (s) {    /* Manager */
      if (peer) {
         const char *id = astman_get_header(m, "ActionID");

         astman_append(s, "Response: Success\r\n");
         if (!ast_strlen_zero(id))
            astman_append(s, "ActionID: %s\r\n", id);
      } else {
         snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
         astman_send_error(s, m, cbuf);
         return CLI_SUCCESS;
      }
   }
   if (peer && type==0 ) { /* Normal listing */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      ast_cli(fd, "\n\n");
      ast_cli(fd, "  * Name       : %s\n", peer->name);
      if (realtimepeers) { /* Realtime is enabled */
         ast_cli(fd, "  Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
      }
      ast_cli(fd, "  Secret       : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  MD5Secret    : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"<Not set>":"<Set>");
      for (auth = peer->auth; auth; auth = auth->next) {
         ast_cli(fd, "  Realm-auth   : Realm %-15.15s User %-10.20s ", auth->realm, auth->username);
         ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>"));
      }
      ast_cli(fd, "  Context      : %s\n", peer->context);
      ast_cli(fd, "  Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
      ast_cli(fd, "  Language     : %s\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         ast_cli(fd, "  Accountcode  : %s\n", peer->accountcode);
      ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(peer->amaflags));
      ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
      ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         ast_cli(fd, "  FromUser     : %s\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         ast_cli(fd, "  FromDomain   : %s Port %d\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
      ast_cli(fd, "  Callgroup    : ");
      print_group(fd, peer->callgroup, 0);
      ast_cli(fd, "  Pickupgroup  : ");
      print_group(fd, peer->pickupgroup, 0);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_cli(fd, "  MOH Suggest  : %s\n", peer->mohsuggest);
      ast_cli(fd, "  Mailbox      : %s\n", mailbox_str->str);
      ast_cli(fd, "  VM Extension : %s\n", peer->vmexten);
      ast_cli(fd, "  LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
      ast_cli(fd, "  Call limit   : %d\n", peer->call_limit);
      ast_cli(fd, "  Max forwards : %d\n", peer->maxforwards);
      if (peer->busy_level)
         ast_cli(fd, "  Busy level   : %d\n", peer->busy_level);
      ast_cli(fd, "  Dynamic      : %s\n", AST_CLI_YESNO(peer->host_dynamic));
      ast_cli(fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
      ast_cli(fd, "  MaxCallBR    : %d kbps\n", peer->maxcallbitrate);
      ast_cli(fd, "  Expire       : %ld\n", ast_sched_when(sched, peer->expire));
      ast_cli(fd, "  Insecure     : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      ast_cli(fd, "  Force rport  : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)));
      ast_cli(fd, "  ACL          : %s\n", AST_CLI_YESNO(peer->ha != NULL));
      ast_cli(fd, "  DirectMedACL : %s\n", AST_CLI_YESNO(peer->directmediaha != NULL));
      ast_cli(fd, "  T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 EC mode : %s\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 MaxDtgrm: %d\n", peer->t38_maxdatagram);
      ast_cli(fd, "  DirectMedia  : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
      ast_cli(fd, "  PromiscRedir : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
      ast_cli(fd, "  User=Phone   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
      ast_cli(fd, "  Video Support: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS)));
      ast_cli(fd, "  Text Support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
      ast_cli(fd, "  Ign SDP ver  : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
      ast_cli(fd, "  Trust RPID   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
      ast_cli(fd, "  Send RPID    : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
      ast_cli(fd, "  Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
      ast_cli(fd, "  Overlap dial : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
      if (peer->outboundproxy)
         ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
                     peer->outboundproxy->force ? "(forced)" : "");

      /* - is enumerated */
      ast_cli(fd, "  DTMFmode     : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      ast_cli(fd, "  Timer T1     : %d\n", peer->timer_t1);
      ast_cli(fd, "  Timer B      : %d\n", peer->timer_b);
      ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
      ast_cli(fd, "  Addr->IP     : %s\n", ast_sockaddr_stringify(&peer->addr));
      ast_cli(fd, "  Defaddr->IP  : %s\n", ast_sockaddr_stringify(&peer->defaddr));
      ast_cli(fd, "  Prim.Transp. : %s\n", get_transport(peer->socket.type));
      ast_cli(fd, "  Allowed.Trsp : %s\n", get_transport_list(peer->transports));
      if (!ast_strlen_zero(sip_cfg.regcontext))
         ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
      ast_cli(fd, "  Def. Username: %s\n", peer->username);
      ast_cli(fd, "  SIP Options  : ");
      if (peer->sipoptions) {
         int lastoption = -1;
         for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
            if (sip_options[x].id != lastoption) {
               if (peer->sipoptions & sip_options[x].id)
                  ast_cli(fd, "%s ", sip_options[x].text);
               lastoption = x;
            }
         }
      } else
         ast_cli(fd, "(none)");

      ast_cli(fd, "\n");
      ast_cli(fd, "  Codecs       : ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      ast_cli(fd, "%s\n", codec_buf);
      ast_cli(fd, "  Codec Order  : (");
      print_codec_to_cli(fd, &peer->prefs);
      ast_cli(fd, ")\n");

      ast_cli(fd, "  Auto-Framing :  %s \n", AST_CLI_YESNO(peer->autoframing));
      ast_cli(fd, "  100 on REG   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_REGISTERTRYING)));
      ast_cli(fd, "  Status       : ");
      peer_status(peer, status, sizeof(status));
      ast_cli(fd, "%s\n", status);
      ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
      ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
      ast_cli(fd, "  Qualify Freq : %d ms\n", peer->qualifyfreq);
      if (peer->chanvars) {
         ast_cli(fd, "  Variables    :\n");
         for (v = peer->chanvars ; v ; v = v->next)
            ast_cli(fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(fd, "  Sess-Timers  : %s\n", stmode2str(peer->stimer.st_mode_oper));
      ast_cli(fd, "  Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
      ast_cli(fd, "  Sess-Expires : %d secs\n", peer->stimer.st_max_se);
      ast_cli(fd, "  Min-Sess     : %d secs\n", peer->stimer.st_min_se);
      ast_cli(fd, "  RTP Engine   : %s\n", peer->engine);
      ast_cli(fd, "  Parkinglot   : %s\n", peer->parkinglot);
      ast_cli(fd, "  Use Reason   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
      ast_cli(fd, "\n");
      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
   } else  if (peer && type == 1) { /* manager listing */
      char buffer[256];
      struct ast_str *mailbox_str = ast_str_alloca(512);
      astman_append(s, "Channeltype: SIP\r\n");
      astman_append(s, "ObjectName: %s\r\n", peer->name);
      astman_append(s, "ChanObjectType: peer\r\n");
      astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
      astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
      astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
      astman_append(s, "Context: %s\r\n", peer->context);
      astman_append(s, "Language: %s\r\n", peer->language);
      if (!ast_strlen_zero(peer->accountcode))
         astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
      astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags));
      astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         astman_append(s, "SIP-FromDomain: %s\r\nSip-FromDomain-Port: %d\r\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
      astman_append(s, "Callgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
      astman_append(s, "Pickupgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
      astman_append(s, "MOHSuggest: %s\r\n", peer->mohsuggest);
      peer_mailboxes_to_str(&mailbox_str, peer);
      astman_append(s, "VoiceMailbox: %s\r\n", mailbox_str->str);
      astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
      astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
      astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
      astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
      astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
      astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
      astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
      astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
      astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
      astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
      astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      astman_append(s, "SIP-Forcerport: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT)?"Y":"N"));
      astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
      astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
      astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
      astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38Support: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38EC: %s\r\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      astman_append(s, "SIP-T.38MaxDtgrm: %d\r\n", peer->t38_maxdatagram);
      astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
      astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresher2str(peer->stimer.st_ref));
      astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
      astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
      astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);

      /* - is enumerated */
      astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      astman_append(s, "ToHost: %s\r\n", peer->tohost);
      astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", ast_sockaddr_stringify_addr(&peer->addr), ast_sockaddr_port(&peer->addr));
      astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_sockaddr_stringify_addr(&peer->defaddr), ast_sockaddr_port(&peer->defaddr));
      astman_append(s, "Default-Username: %s\r\n", peer->username);
      if (!ast_strlen_zero(sip_cfg.regcontext))
         astman_append(s, "RegExtension: %s\r\n", peer->regexten);
      astman_append(s, "Codecs: ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
      astman_append(s, "%s\r\n", codec_buf);
      astman_append(s, "CodecOrder: ");
      pref = &peer->prefs;
      for(x = 0; x < 64 ; x++) {
         codec = ast_codec_pref_index(pref, x);
         if (!codec)
            break;
         astman_append(s, "%s", ast_getformatname(codec));
         if (x < 63 && ast_codec_pref_index(pref, x+1))
            astman_append(s, ",");
      }

      astman_append(s, "\r\n");
      astman_append(s, "Status: ");
      peer_status(peer, status, sizeof(status));
      astman_append(s, "%s\r\n", status);
      astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
      astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
      astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
      astman_append(s, "Parkinglot: %s\r\n", peer->parkinglot);
      if (peer->chanvars) {
         for (v = peer->chanvars ; v ; v = v->next) {
            astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
         }
      }
      astman_append(s, "SIP-Use-Reason-Header : %s\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");

      peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer");

   } else {
      ast_cli(fd, "Peer %s not found.\n", argv[3]);
      ast_cli(fd, "\n");
   }

   return CLI_SUCCESS;
}

static char * _sip_show_peers	(	int	fd,
		int *	total,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)			[static]

Execute sip show peers command.

Definition at line 15515 of file chan_sip.c.

References ao2_container_count(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_calloc, ast_check_realtime(), ast_cli(), ast_copy_string(), ast_free, ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, FORMAT, FORMAT2, id, name, peer_status(), peercomparefunc(), status, TRUE, and unref_peer().

Referenced by manager_sip_show_peers(), and sip_show_peers().

{
   regex_t regexbuf;
   int havepattern = FALSE;
   struct sip_peer *peer;
   struct ao2_iterator i;
   
/* the last argument is left-aligned, so we don't need a size anyways */
#define FORMAT2 "%-25.25s  %-39.39s %-3.3s %-10.10s %-3.3s %-8s %-10s %s\n"
#define FORMAT  "%-25.25s  %-39.39s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"

   char name[256];
   int total_peers = 0;
   int peers_mon_online = 0;
   int peers_mon_offline = 0;
   int peers_unmon_offline = 0;
   int peers_unmon_online = 0;
   const char *id;
   char idtext[256] = "";
   int realtimepeers;
   int objcount = ao2_container_count(peers);
   struct sip_peer **peerarray;
   int k;
   
   
   realtimepeers = ast_check_realtime("sippeers");
   peerarray = ast_calloc(sizeof(struct sip_peer *), objcount);

   if (s) { /* Manager - get ActionID */
      id = astman_get_header(m, "ActionID");
      if (!ast_strlen_zero(id))
         snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
   }

   switch (argc) {
   case 5:
      if (!strcasecmp(argv[3], "like")) {
         if (regcomp(&regexbuf, argv[4], REG_EXTENDED | REG_NOSUB))
            return CLI_SHOWUSAGE;
         havepattern = TRUE;
      } else
         return CLI_SHOWUSAGE;
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (!s) /* Normal list */
      ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Forcerport", "ACL", "Port", "Status", (realtimepeers ? "Realtime" : ""));
   

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {  
      ao2_lock(peer);

      if (!(peer->type & SIP_TYPE_PEER)) {
         ao2_unlock(peer);
         unref_peer(peer, "unref peer because it's actually a user");
         continue;
      }

      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         objcount--;
         ao2_unlock(peer);
         unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      peerarray[total_peers++] = peer;
      ao2_unlock(peer);
   }
   ao2_iterator_destroy(&i);
   
   qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);

   for(k=0; k < total_peers; k++) {
      char status[20] = "";
      char srch[2000];
      char pstatus;
      peer = peerarray[k];
      
      ao2_lock(peer);
      if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
         ao2_unlock(peer);
         unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      if (!ast_strlen_zero(peer->username) && !s)
         snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
      else
         ast_copy_string(name, peer->name, sizeof(name));
      
      pstatus = peer_status(peer, status, sizeof(status));
      if (pstatus == 1)
         peers_mon_online++;
      else if (pstatus == 0)
         peers_mon_offline++;
      else {
         if (ast_sockaddr_isnull(&peer->addr) ||
             !ast_sockaddr_port(&peer->addr)) {
            peers_unmon_offline++;
         } else {
            peers_unmon_online++;
         }
      }

      snprintf(srch, sizeof(srch), FORMAT, name,
         ast_sockaddr_isnull(&peer->addr) ? "(Unspecified)" : ast_sockaddr_stringify_addr(&peer->addr),
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",  /* permit/deny */
         ast_sockaddr_isnull(&peer->addr) ? 0 : ast_sockaddr_port(&peer->addr), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");

      if (!s)  {/* Normal CLI list */
         ast_cli(fd, FORMAT, name,
         ast_sockaddr_isnull(&peer->addr) ? "(Unspecified)" : ast_sockaddr_stringify_addr(&peer->addr),
         peer->host_dynamic ? " D " : "   ",    /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? " N " : "   ", /* NAT=yes? */
         peer->ha ? " A " : "   ",       /* permit/deny */
         ast_sockaddr_isnull(&peer->addr) ? 0 : ast_sockaddr_port(&peer->addr), status,
         realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");
      } else { /* Manager format */
         /* The names here need to be the same as other channels */
         astman_append(s,
         "Event: PeerEntry\r\n%s"
         "Channeltype: SIP\r\n"
         "ObjectName: %s\r\n"
         "ChanObjectType: peer\r\n" /* "peer" or "user" */
         "IPaddress: %s\r\n"
         "IPport: %d\r\n"
         "Dynamic: %s\r\n"
         "Forcerport: %s\r\n"
         "VideoSupport: %s\r\n"
         "TextSupport: %s\r\n"
         "ACL: %s\r\n"
         "Status: %s\r\n"
         "RealtimeDevice: %s\r\n\r\n",
         idtext,
         peer->name,
         ast_sockaddr_isnull(&peer->addr) ? "-none-" : ast_sockaddr_stringify(&peer->addr),
         ast_sockaddr_isnull(&peer->addr) ? 0 : ast_sockaddr_port(&peer->addr),
         peer->host_dynamic ? "yes" : "no",  /* Dynamic or not? */
         ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "yes" : "no",  /* NAT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no",  /* VIDEOSUPPORT=yes? */
         ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no",   /* TEXTSUPPORT=yes? */
         peer->ha ? "yes" : "no",       /* permit/deny */
         status,
         realtimepeers ? (peer->is_realtime ? "yes":"no") : "no");
      }
      ao2_unlock(peer);
      unref_peer(peer, "toss iterator peer ptr");
   }
   
   if (!s)
      ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
              total_peers, peers_mon_online, peers_mon_offline, peers_unmon_online, peers_unmon_offline);

   if (havepattern)
      regfree(&regexbuf);

   if (total)
      *total = total_peers;
   
   ast_free(peerarray);
   
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static void * _sip_tcp_helper_thread	(	struct sip_pvt *	pvt,
		struct ast_tcptls_session_instance *	tcptls_session
	)			[static]

SIP TCP thread management function This function reads from the socket, parses the packet into a request.

Todo:

XXX If there's no Content-Length or if the content-length and what we receive is not the same - we should generate an error

Definition at line 2428 of file chan_sip.c.

References ao2_lock, ao2_ref, ao2_t_find, ao2_t_ref, ao2_t_unlink, ao2_unlock, ast_debug, AST_LIST_REMOVE_HEAD, ast_log(), ast_mutex_lock, ast_mutex_unlock, ast_poll, ast_str_append(), ast_str_buffer(), ast_str_create(), ast_str_reset(), ast_tcptls_client_start(), ast_tcptls_server_write(), cleanup(), ast_tcptls_session_instance::client, copy_request(), deinit_req(), errno, ast_tcptls_session_instance::f, ast_tcptls_session_instance::fd, get_header(), handle_request_do(), len(), ast_tcptls_session_instance::lock, LOG_ERROR, LOG_WARNING, MIN, OBJ_POINTER, ast_tcptls_session_instance::parent, parse_request(), ast_tcptls_session_instance::remote_address, set_socket_transport(), sip_threadinfo_create(), and ast_tcptls_session_instance::ssl.

Referenced by sip_tcp_worker_fn().

{
   int res, cl;
   struct sip_request req = { 0, } , reqcpy = { 0, };
   struct sip_threadinfo *me = NULL;
   char buf[1024] = "";
   struct pollfd fds[2] = { { 0 }, { 0 }, };
   struct ast_tcptls_session_args *ca = NULL;

   /* If this is a server session, then the connection has already been setup,
    * simply create the threadinfo object so we can access this thread for writing.
    * 
    * if this is a client connection more work must be done.
    * 1. We own the parent session args for a client connection.  This pointer needs
    *    to be held on to so we can decrement it's ref count on thread destruction.
    * 2. The threadinfo object was created before this thread was launched, however
    *    it must be found within the threadt table.
    * 3. Last, the tcptls_session must be started.
    */
   if (!tcptls_session->client) {
      if (!(me = sip_threadinfo_create(tcptls_session, tcptls_session->ssl ? SIP_TRANSPORT_TLS : SIP_TRANSPORT_TCP))) {
         goto cleanup;
      }
      ao2_t_ref(me, +1, "Adding threadinfo ref for tcp_helper_thread");
   } else {
      struct sip_threadinfo tmp = {
         .tcptls_session = tcptls_session,
      };

      if ((!(ca = tcptls_session->parent)) ||
         (!(me = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread"))) ||
         (!(tcptls_session = ast_tcptls_client_start(tcptls_session)))) {
         goto cleanup;
      }
   }

   me->threadid = pthread_self();
   ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

   /* set up pollfd to watch for reads on both the socket and the alert_pipe */
   fds[0].fd = tcptls_session->fd;
   fds[1].fd = me->alert_pipe[0];
   fds[0].events = fds[1].events = POLLIN | POLLPRI;

   if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
      goto cleanup;
   }
   if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET))) {
      goto cleanup;
   }

   for (;;) {
      struct ast_str *str_save;

      res = ast_poll(fds, 2, -1); /* polls for both socket and alert_pipe */
      if (res < 0) {
         ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
         goto cleanup;
      }

      /* handle the socket event, check for both reads from the socket fd,
       * and writes from alert_pipe fd */
      if (fds[0].revents) { /* there is data on the socket to be read */

         fds[0].revents = 0;

         /* clear request structure */
         str_save = req.data;
         memset(&req, 0, sizeof(req));
         req.data = str_save;
         ast_str_reset(req.data);

         str_save = reqcpy.data;
         memset(&reqcpy, 0, sizeof(reqcpy));
         reqcpy.data = str_save;
         ast_str_reset(reqcpy.data);

         memset(buf, 0, sizeof(buf));

         if (tcptls_session->ssl) {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TLS);
            req.socket.port = htons(ourport_tls);
         } else {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TCP);
            req.socket.port = htons(ourport_tcp);
         }
         req.socket.fd = tcptls_session->fd;

         /* Read in headers one line at a time */
         while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
            ast_mutex_lock(&tcptls_session->lock);
            if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
               ast_mutex_unlock(&tcptls_session->lock);
               goto cleanup;
            }
            ast_mutex_unlock(&tcptls_session->lock);
            if (me->stop) {
                goto cleanup;
            }
            ast_str_append(&req.data, 0, "%s", buf);
            req.len = req.data->used;
         }
         copy_request(&reqcpy, &req);
         parse_request(&reqcpy);
         /* In order to know how much to read, we need the content-length header */
         if (sscanf(get_header(&reqcpy, "Content-Length"), "%30d", &cl)) {
            while (cl > 0) {
               size_t bytes_read;
               ast_mutex_lock(&tcptls_session->lock);
               if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, cl), tcptls_session->f))) {
                  ast_mutex_unlock(&tcptls_session->lock);
                  goto cleanup;
               }
               buf[bytes_read] = '\0';
               ast_mutex_unlock(&tcptls_session->lock);
               if (me->stop) {
                  goto cleanup;
               }
               cl -= strlen(buf);
               ast_str_append(&req.data, 0, "%s", buf);
               req.len = req.data->used;
            }
         }
         /*! \todo XXX If there's no Content-Length or if the content-length and what
               we receive is not the same - we should generate an error */

         req.socket.tcptls_session = tcptls_session;
         handle_request_do(&req, &tcptls_session->remote_address);
      }

      if (fds[1].revents) { /* alert_pipe indicates there is data in the send queue to be sent */
         enum sip_tcptls_alert alert;
         struct tcptls_packet *packet;

         fds[1].revents = 0;

         if (read(me->alert_pipe[0], &alert, sizeof(alert)) == -1) {
            ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
            continue;
         }

         switch (alert) {
         case TCPTLS_ALERT_STOP:
            goto cleanup;
         case TCPTLS_ALERT_DATA:
            ao2_lock(me);
            if (!(packet = AST_LIST_REMOVE_HEAD(&me->packet_q, entry))) {
               ast_log(LOG_WARNING, "TCPTLS thread alert_pipe indicated packet should be sent, but frame_q is empty");
            } else if (ast_tcptls_server_write(tcptls_session, ast_str_buffer(packet->data), packet->len) == -1) {
               ast_log(LOG_WARNING, "Failure to write to tcp/tls socket\n");
            }

            if (packet) {
               ao2_t_ref(packet, -1, "tcptls packet sent, this is no longer needed");
            }
            ao2_unlock(me);
            break;
         default:
            ast_log(LOG_ERROR, "Unknown tcptls thread alert '%d'\n", alert);
         }
      }
   }

   ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

cleanup:
   if (me) {
      ao2_t_unlink(threadt, me, "Removing tcptls helper thread, thread is closing");
      ao2_t_ref(me, -1, "Removing tcp_helper_threads threadinfo ref");
   }
   deinit_req(&reqcpy);
   deinit_req(&req);

   /* if client, we own the parent session arguments and must decrement ref */
   if (ca) {
      ao2_t_ref(ca, -1, "closing tcptls thread, getting rid of client tcptls_session arguments");
   }

   if (tcptls_session) {
      ast_mutex_lock(&tcptls_session->lock);
      if (tcptls_session->f) {
         fclose(tcptls_session->f);
         tcptls_session->f = NULL;
      }
      if (tcptls_session->fd != -1) {
         close(tcptls_session->fd);
         tcptls_session->fd = -1;
      }
      tcptls_session->parent = NULL;
      ast_mutex_unlock(&tcptls_session->lock);

      ao2_ref(tcptls_session, -1);
      tcptls_session = NULL;
   }
   return NULL;
}

static void add_blank

(

struct sip_request *

req

)

[static]

add a blank line if no body

Definition at line 3815 of file chan_sip.c.

References ast_str_append(), and ast_str_strlen().

Referenced by send_request(), and send_response().

{
   if (!req->lines) {
      /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
      ast_str_append(&req->data, 0, "\r\n");
      req->len = ast_str_strlen(req->data);
   }
}

static void add_cc_call_info_to_response	(	struct sip_pvt *	p,
		struct sip_request *	resp
	)			[static]

Definition at line 10819 of file chan_sip.c.

References add_header(), ao2_ref, ast_copy_string(), ast_log(), ast_str_alloca, ast_str_buffer(), ast_str_set(), ast_strlen_zero(), find_sip_cc_agent_by_original_callid(), generate_uri(), LOG_WARNING, and ast_cc_agent::private_data.

Referenced by __transmit_response(), and transmit_response_with_sdp().

{
   char uri[SIPBUFSIZE];
   struct ast_str *header = ast_str_alloca(SIPBUFSIZE);
   struct ast_cc_agent *agent = find_sip_cc_agent_by_original_callid(p);
   struct sip_cc_agent_pvt *agent_pvt;

   if (!agent) {
      /* Um, what? How could the SIP_OFFER_CC flag be set but there not be an
       * agent? Oh well, we'll just warn and return without adding the header.
       */
      ast_log(LOG_WARNING, "Can't find SIP CC agent for call '%s' even though OFFER_CC flag was set?\n", p->callid);
      return;
   }

   agent_pvt = agent->private_data;

   if (!ast_strlen_zero(agent_pvt->subscribe_uri)) {
      ast_copy_string(uri, agent_pvt->subscribe_uri, sizeof(uri));
   } else {
      generate_uri(p, uri, sizeof(uri));
      ast_copy_string(agent_pvt->subscribe_uri, uri, sizeof(agent_pvt->subscribe_uri));
   }
   /* XXX Hardcode "NR" as the m reason for now. This should perhaps be changed
    * to be more accurate. This parameter has no bearing on the actual operation
    * of the feature; it's just there for informational purposes.
    */
   ast_str_set(&header, 0, "<%s>;purpose=call-completion;m=%s", uri, "NR");
   add_header(resp, "Call-Info", ast_str_buffer(header));
   ao2_ref(agent, -1);
}

static void add_codec_to_sdp	(	const struct sip_pvt *	p,
		format_t	codec,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 10129 of file chan_sip.c.

References ast_codec_pref_getsize(), AST_FORMAT_G719, AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_ILBC, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), AST_RTP_OPT_G726_NONSTANDARD, ast_str_append(), ast_test_flag, ast_verbose, ast_format_list::cur_ms, and ast_rtp_codecs::pref.

Referenced by add_sdp().

{
   int rtp_code;
   struct ast_format_list fmt;


   if (debug)
      ast_verbose("Adding codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));
   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
      return;

   if (p->rtp) {
      struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
      fmt = ast_codec_pref_getsize(pref, codec);
   } else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
      return;
   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(1, codec,
                     ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
             ast_rtp_lookup_sample_rate2(1, codec));

   switch (codec) {
   case AST_FORMAT_G729A:
      /* Indicate that we don't support VAD (G.729 annex B) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
      break;
   case AST_FORMAT_G723_1:
      /* Indicate that we don't support VAD (G.723.1 annex A) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
      break;
   case AST_FORMAT_ILBC:
      /* Add information about us using only 20/30 ms packetization */
      ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
      break;
   case AST_FORMAT_SIREN7:
      /* Indicate that we only expect 32Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
      break;
   case AST_FORMAT_SIREN14:
      /* Indicate that we only expect 48Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
      break;
   case AST_FORMAT_G719:
      /* Indicate that we only expect 64Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
      break;
   }

   if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
      *min_packet_size = fmt.cur_ms;

   /* Our first codec packetization processed cannot be zero */
   if ((*min_packet_size)==0 && fmt.cur_ms)
      *min_packet_size = fmt.cur_ms;
}

static int add_content	(	struct sip_request *	req,
		const char *	line
	)			[static]

Add content (not header) to SIP message.

Definition at line 9104 of file chan_sip.c.

References ast_log(), ast_str_append(), and LOG_WARNING.

Referenced by add_digit(), add_sdp(), add_text(), add_vidupdate(), transmit_cc_notify(), transmit_invite(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), and transmit_state_notify().

{
   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more content when the content has been finalized\n");
      return -1;
   }

   ast_str_append(&req->content, 0, "%s", line);
   return 0;
}

static int add_digit	(	struct sip_request *	req,
		char	digit,
		unsigned int	duration,
		int	mode
	)			[static]

Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.

Definition at line 9993 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_info_with_digit().

{
   char tmp[256];
   int event;
   if (mode) {
      /* Application/dtmf short version used by some implementations */
      if (digit == '*')
         event = 10;
      else if (digit == '#')
         event = 11;
      else if ((digit >= 'A') && (digit <= 'D'))
         event = 12 + digit - 'A';
      else
         event = atoi(&digit);
      snprintf(tmp, sizeof(tmp), "%d\r\n", event);
      add_header(req, "Content-Type", "application/dtmf");
      add_content(req, tmp);
   } else {
      /* Application/dtmf-relay as documented by Cisco */
      snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
      add_header(req, "Content-Type", "application/dtmf-relay");
      add_content(req, tmp);
   }
   return 0;
}

static void add_diversion_header	(	struct sip_request *	req,
		struct sip_pvt *	pvt
	)			[static]

Add "Diversion" header to outgoing message.

We need to add a Diversion header if the owner channel of this dialog has redirecting information associated with it.

Parameters:

	req	The request/response to which we will add the header
	pvt	The sip_pvt which represents the call-leg

Definition at line 11201 of file chan_sip.c.

References add_header(), ast_sockaddr_stringify_host(), ast_strlen_zero(), and sip_reason_code_to_str().

Referenced by transmit_invite(), and update_redirecting().

{
   const char *diverting_number;
   const char *diverting_name;
   const char *reason;
   char header_text[256];

   if (!pvt->owner) {
      return;
   }

   diverting_number = pvt->owner->redirecting.from.number.str;
   if (!pvt->owner->redirecting.from.number.valid
      || ast_strlen_zero(diverting_number)) {
      return;
   }

   reason = sip_reason_code_to_str(pvt->owner->redirecting.reason);

   /* We at least have a number to place in the Diversion header, which is enough */
   diverting_name = pvt->owner->redirecting.from.name.str;
   if (!pvt->owner->redirecting.from.name.valid
      || ast_strlen_zero(diverting_name)) {
      snprintf(header_text, sizeof(header_text), "<sip:%s@%s>;reason=%s", diverting_number,
            ast_sockaddr_stringify_host(&pvt->ourip), reason);
   } else {
      snprintf(header_text, sizeof(header_text), "\"%s\" <sip:%s@%s>;reason=%s",
            diverting_name, diverting_number,
            ast_sockaddr_stringify_host(&pvt->ourip), reason);
   }

   add_header(req, "Diversion", header_text);
}

static int add_header	(	struct sip_request *	req,
		const char *	var,
		const char *	value
	)			[static]

Add header to SIP message.

Definition at line 9032 of file chan_sip.c.

References ast_log(), ast_str_append(), ast_str_strlen(), find_alias(), LOG_WARNING, and sip_cfg.

{
   if (req->headers == SIP_MAX_HEADERS) {
      ast_log(LOG_WARNING, "Out of SIP header space\n");
      return -1;
   }

   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
      return -1;
   }

   if (sip_cfg.compactheaders) {
      var = find_alias(var, var);
   }

   ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
   req->header[req->headers] = req->len;

   req->len = ast_str_strlen(req->data);
   req->headers++;

   return 0;   
}

static int add_header_max_forwards	(	struct sip_pvt *	dialog,
		struct sip_request *	req
	)			[static]

Add 'Max-Forwards' header to SIP message.

Definition at line 9058 of file chan_sip.c.

References add_header(), ast_channel_trylock, ast_channel_unlock, pbx_builtin_getvar_helper(), sip_pvt_lock, and sip_pvt_unlock.

Referenced by initreqprep(), reqprep(), and transmit_register().

{
   char clen[10];
   const char *max = NULL;

   /* deadlock avoidance */
   while (dialog->owner && ast_channel_trylock(dialog->owner)) {
      sip_pvt_unlock(dialog);
      usleep(1);
      sip_pvt_lock(dialog);
   }

   if (dialog->owner) {
      max = pbx_builtin_getvar_helper(dialog->owner, "SIP_MAX_FORWARDS");
      ast_channel_unlock(dialog->owner);
   }

   /* The channel variable overrides the peer/channel value */
   if (max == NULL) {
      snprintf(clen, sizeof(clen), "%d", dialog->maxforwards);
   }
   return add_header(req, "Max-Forwards", max != NULL ? max : clen);
}

static void add_noncodec_to_sdp	(	const struct sip_pvt *	p,
		int	format,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug
	)			[static]

Add RFC 2833 DTMF offer to SDP.

Definition at line 10267 of file chan_sip.c.

References ast_rtp_codecs_payload_code(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), and ast_verbose.

Referenced by add_sdp().

{
   int rtp_code;

   if (debug)
      ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0));
   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(0, format, 0),
             ast_rtp_lookup_sample_rate2(0, format));
   if (format == AST_RTP_DTMF)   /* Indicate we support DTMF and FLASH... */
      ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}

static void add_peer_mailboxes	(	struct sip_peer *	peer,
		const char *	value
	)			[static]

Todo:

document this function

Definition at line 25536 of file chan_sip.c.

References ast_calloc, AST_LIST_INSERT_TAIL, AST_LIST_TRAVERSE, ast_strdupa, ast_strlen_zero(), context, mailbox, mbox(), S_OR, and strsep().

Referenced by build_peer().

{
   char *next, *mbox, *context;

   next = ast_strdupa(value);

   while ((mbox = context = strsep(&next, ","))) {
      struct sip_mailbox *mailbox;
      int duplicate = 0;

      strsep(&context, "@");

      if (ast_strlen_zero(mbox)) {
         continue;
      }

      /* Check whether the mailbox is already in the list */
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         if (!strcmp(mailbox->mailbox, mbox) && !strcmp(S_OR(mailbox->context, ""), S_OR(context, ""))) {
            duplicate = 1;
            mailbox->delme = 0;
            break;
         }
      }
      if (duplicate) {
         continue;
      }

      if (!(mailbox = ast_calloc(1, sizeof(*mailbox) + strlen(mbox) + strlen(S_OR(context, ""))))) {
         continue;
      }

      if (!ast_strlen_zero(context)) {
         mailbox->context = mailbox->mailbox + strlen(mbox) + 1;
         strcpy(mailbox->context, context); /* SAFE */
      }
      strcpy(mailbox->mailbox, mbox); /* SAFE */

      AST_LIST_INSERT_TAIL(&peer->mailboxes, mailbox, entry);
   }
}

static void add_peer_mwi_subs

(

struct sip_peer *

peer

)

[static]

Definition at line 22883 of file chan_sip.c.

References AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, ast_event_subscribe(), AST_LIST_TRAVERSE, mailbox, mwi_event_cb(), and S_OR.

Referenced by build_peer(), and handle_request_subscribe().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      mailbox->event_sub = ast_event_subscribe(AST_EVENT_MWI, mwi_event_cb, "SIP mbox event", peer,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
   }
}

static struct sip_auth * add_realm_authentication	(	struct sip_auth *	authlist,
		const char *	configuration,
		int	lineno
	)			[static, read]

Add realm authentication in list.

Definition at line 25349 of file chan_sip.c.

References ast_calloc, ast_copy_string(), ast_debug, ast_log(), ast_strlen_zero(), ast_verb, LOG_WARNING, and secret.

Referenced by build_peer(), and reload_config().

{
   char authcopy[256];
   char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL;
   struct sip_auth *a, *b, *auth;

   if (ast_strlen_zero(configuration))
      return authlist;

   ast_debug(1, "Auth config ::  %s\n", configuration);

   ast_copy_string(authcopy, configuration, sizeof(authcopy));
   username = authcopy;

   /* split user[:secret] and relm */
   realm = strrchr(username, '@');
   if (realm)
      *realm++ = '\0';
   if (ast_strlen_zero(username) || ast_strlen_zero(realm)) {
      ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno);
      return authlist;
   }

   /* parse username at ':' for secret, or '#" for md5secret */
   if ((secret = strchr(username, ':'))) {
      *secret++ = '\0';
   } else if ((md5secret = strchr(username, '#'))) {
      *md5secret++ = '\0';
   }

   if (!(auth = ast_calloc(1, sizeof(*auth))))
      return authlist;

   ast_copy_string(auth->realm, realm, sizeof(auth->realm));
   ast_copy_string(auth->username, username, sizeof(auth->username));
   if (secret)
      ast_copy_string(auth->secret, secret, sizeof(auth->secret));
   if (md5secret)
      ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret));

   /* find the end of the list */
   for (b = NULL, a = authlist; a ; b = a, a = a->next)
      ;
   if (b)
      b->next = auth;   /* Add structure add end of list */
   else
      authlist = auth;

   ast_verb(3, "Added authentication for realm %s\n", realm);

   return authlist;

}

static void add_route	(	struct sip_request *	req,
		struct sip_route *	route
	)			[static]

Add route header into request per learned route.

Definition at line 9215 of file chan_sip.c.

References add_header(), and ast_copy_string().

Referenced by initreqprep(), and reqprep().

{
   char r[SIPBUFSIZE*2], *p;
   int n, rem = sizeof(r);

   if (!route)
      return;

   p = r;
   for (;route ; route = route->next) {
      n = strlen(route->hop);
      if (rem < n+3) /* we need room for ",<route>" */
         break;
      if (p != r) {  /* add a separator after fist route */
         *p++ = ',';
         --rem;
      }
      *p++ = '<';
      ast_copy_string(p, route->hop, rem); /* cannot fail */
      p += n;
      *p++ = '>';
      rem -= (n+2);
   }
   *p = '\0';
   add_header(req, "Route", r);
}

static int add_rpid	(	struct sip_request *	req,
		struct sip_pvt *	p
	)			[static]

Add Remote-Party-ID header to SIP message.

Precondition:

if p->owner exists, it must be locked

Definition at line 10023 of file chan_sip.c.

References add_header(), ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_ALLOWED_NETWORK_NUMBER, AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRES_NUMBER_NOT_AVAILABLE, AST_PRES_PROHIB_NETWORK_NUMBER, AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, AST_PRES_RESTRICTION, ast_sockaddr_stringify_host(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_set(), ast_strlen_zero(), ast_test_flag, ast_uri_encode(), and S_OR.

Referenced by __transmit_response(), transmit_invite(), transmit_reinvite_with_sdp(), transmit_response_with_sdp(), and update_connectedline().

{
   struct ast_str *tmp = ast_str_alloca(256);
   char tmp2[256];
   char *lid_num = NULL;
   char *lid_name = NULL;
   int lid_pres;
   const char *fromdomain;
   const char *privacy = NULL;
   const char *screen = NULL;
   const char *anonymous_string = "\"Anonymous\" <anonymous@anonymous.invalid>";

   if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
      return 0;
   }

   if (p->owner && p->owner->connected.id.number.valid
      && p->owner->connected.id.number.str) {
      lid_num = p->owner->connected.id.number.str;
   }
   if (p->owner && p->owner->connected.id.name.valid
      && p->owner->connected.id.name.str) {
      lid_name = p->owner->connected.id.name.str;
   }
   lid_pres = (p->owner) ? ast_party_id_presentation(&p->owner->connected.id) : AST_PRES_NUMBER_NOT_AVAILABLE;

   if (ast_strlen_zero(lid_num))
      return 0;
   if (ast_strlen_zero(lid_name))
      lid_name = lid_num;
   fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host(&p->ourip));

   lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), 1);

   if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
      if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
         ast_str_set(&tmp, -1, "%s", anonymous_string);
      } else {
         ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
      }
      add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
   } else {
      ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "calling" : "called");

      switch (lid_pres) {
      case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
      case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
         privacy = "off";
         screen = "no";
         break;
      case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
      case AST_PRES_ALLOWED_NETWORK_NUMBER:
         privacy = "off";
         screen = "yes";
         break;
      case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
      case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
         privacy = "full";
         screen = "no";
         break;
      case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
      case AST_PRES_PROHIB_NETWORK_NUMBER:
         privacy = "full";
         screen = "yes";
         break;
      case AST_PRES_NUMBER_NOT_AVAILABLE:
         break;
      default:
         if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
            privacy = "full";
         }
         else
            privacy = "off";
         screen = "no";
         break;
      }

      if (!ast_strlen_zero(privacy) && !ast_strlen_zero(screen)) {
         ast_str_append(&tmp, -1, ";privacy=%s;screen=%s", privacy, screen);
      }

      add_header(req, "Remote-Party-ID", ast_str_buffer(tmp));
   }
   return 0;
}

static enum sip_result add_sdp	(	struct sip_request *	resp,
		struct sip_pvt *	p,
		int	oldsdp,
		int	add_audio,
		int	add_t38
	)			[static]

Add Session Description Protocol message.

If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions without modifying the media session in any way.

Definition at line 10405 of file chan_sip.c.

References add_codec_to_sdp(), add_content(), add_header(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ast_codec_pref_index(), ast_debug, AST_FORMAT_AUDIO_MASK, AST_FORMAT_TEXT_MASK, AST_FORMAT_VIDEO_MASK, ast_getformatname_multiple(), ast_internal_timing_enabled(), ast_log(), ast_random(), AST_RTP_MAX, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_is_ipv4_mapped(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), ast_str_alloca, ast_str_append(), ast_strlen_zero(), AST_T38_RATE_MANAGEMENT_LOCAL_TCF, AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF, ast_test_flag, ast_udptl_get_error_correction_scheme(), ast_udptl_get_local_max_datagram(), ast_udptl_get_us(), ast_verbose, capability, debug, FALSE, get_crypto_attrib(), get_our_media_address(), LOG_WARNING, sip_debug_test_pvt(), t38_get_rate(), TRUE, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, UDPTL_ERROR_CORRECTION_REDUNDANCY, and version.

{
   format_t alreadysent = 0;

   struct ast_sockaddr addr = { {0,} };
   struct ast_sockaddr vaddr = { {0,} };
   struct ast_sockaddr taddr = { {0,} };
   struct ast_sockaddr udptladdr = { {0,} };
   struct ast_sockaddr dest = { {0,} };
   struct ast_sockaddr vdest = { {0,} };
   struct ast_sockaddr tdest = { {0,} };
   struct ast_sockaddr udptldest = { {0,} };

   /* SDP fields */
   char *version =   "v=0\r\n";     /* Protocol version */
   char subject[256];            /* Subject of the session */
   char owner[256];           /* Session owner/creator */
   char connection[256];            /* Connection data */
   char *session_time = "t=0 0\r\n";         /* Time the session is active */
   char bandwidth[256] = "";        /* Max bitrate */
   char *hold = "";
   struct ast_str *m_audio = ast_str_alloca(256);  /* Media declaration line for audio */
   struct ast_str *m_video = ast_str_alloca(256);  /* Media declaration line for video */
   struct ast_str *m_text = ast_str_alloca(256);   /* Media declaration line for text */
   struct ast_str *m_modem = ast_str_alloca(256);  /* Media declaration line for modem */
   struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
   struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
   struct ast_str *a_text = ast_str_alloca(1024);  /* Attributes for text */
   struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
   const char *a_crypto = NULL;
   const char *v_a_crypto = NULL;
   const char *t_a_crypto = NULL;

   format_t x;
   format_t capability = 0;
   int needaudio = FALSE;
   int needvideo = FALSE;
   int needtext = FALSE;
   int debug = sip_debug_test_pvt(p);
   int min_audio_packet_size = 0;
   int min_video_packet_size = 0;
   int min_text_packet_size = 0;

   char codecbuf[SIPBUFSIZE];
   char buf[SIPBUFSIZE];
   char dummy_answer[256];

   /* Set the SDP session name */
   snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);

   if (!p->rtp) {
      ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
      return AST_FAILURE;
   }
   /* XXX We should not change properties in the SIP dialog until
      we have acceptance of the offer if this is a re-invite */

   /* Set RTP Session ID and version */
   if (!p->sessionid) {
      p->sessionid = (int)ast_random();
      p->sessionversion = p->sessionid;
   } else {
      if (oldsdp == FALSE)
         p->sessionversion++;
   }

   if (add_audio) {
      /* Check if we need video in this call */
      if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
         if (p->vrtp) {
            needvideo = TRUE;
            ast_debug(2, "This call needs video offers!\n");
         } else
            ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
      }
      /* Check if we need text in this call */
      if ((p->jointcapability & AST_FORMAT_TEXT_MASK) && !p->notext) {
         if (sipdebug_text)
            ast_verbose("We think we can do text\n");
         if (p->trtp) {
            if (sipdebug_text) {
               ast_verbose("And we have a text rtp object\n");
            }
            needtext = TRUE;
            ast_debug(2, "This call needs text offers! \n");
         } else {
            ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
         }
      }
   }

   get_our_media_address(p, needvideo, needtext, &addr, &vaddr, &taddr, &dest, &vdest, &tdest);

   snprintf(owner, sizeof(owner), "o=%s %d %d IN %s %s\r\n",
       ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner,
       p->sessionid, p->sessionversion,
       (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
         "IP6" : "IP4",
       ast_sockaddr_stringify_addr(&dest));

   snprintf(connection, sizeof(connection), "c=IN %s %s\r\n",
       (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
         "IP6" : "IP4",
       ast_sockaddr_stringify_addr(&dest));

   if (add_audio) {
      capability = p->jointcapability;

      /* XXX note, Video and Text are negated - 'true' means 'no' */
      ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
           p->novideo ? "True" : "False", p->notext ? "True" : "False");
      ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));

      /* Check if we need audio */
      if (capability & AST_FORMAT_AUDIO_MASK)
         needaudio = TRUE;

      if (debug) {
         ast_verbose("Audio is at %s\n", ast_sockaddr_stringify_port(&p->ourip));
      }

      /* Ok, we need video. Let's add what we need for video and set codecs.
         Video is handled differently than audio since we can not transcode. */
      if (needvideo) {
         get_crypto_attrib(p->vsrtp, &v_a_crypto);
         ast_str_append(&m_video, 0, "m=video %d RTP/%s", ast_sockaddr_port(&vdest),
            v_a_crypto ? "SAVP" : "AVP");

         /* Build max bitrate string */
         if (p->maxcallbitrate)
            snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
         if (debug) {
            ast_verbose("Video is at %s\n", ast_sockaddr_stringify(&p->ourip));
         }
      }

      /* Ok, we need text. Let's add what we need for text and set codecs.
         Text is handled differently than audio since we can not transcode. */
      if (needtext) {
         if (sipdebug_text)
            ast_verbose("Lets set up the text sdp\n");
         get_crypto_attrib(p->tsrtp, &t_a_crypto);
         ast_str_append(&m_text, 0, "m=text %d RTP/%s", ast_sockaddr_port(&tdest),
            t_a_crypto ? "SAVP" : "AVP");
         if (debug) {  /* XXX should I use tdest below ? */
            ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&p->ourip));
         }
      }

      /* Start building generic SDP headers */

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      get_crypto_attrib(p->srtp, &a_crypto);
      ast_str_append(&m_audio, 0, "m=audio %d RTP/%s", ast_sockaddr_port(&dest),
         a_crypto ? "SAVP" : "AVP");

      if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
         hold = "a=recvonly\r\n";
      else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
         hold = "a=inactive\r\n";
      else
         hold = "a=sendrecv\r\n";

      /* Now, start adding audio codecs. These are added in this order:
         - First what was requested by the calling channel
         - Then preferences in order from sip.conf device config for this peer/user
         - Then other codecs in capabilities, including video
      */

      /* Prefer the audio codec we were requested to use, first, no matter what
         Note that p->prefcodec can include video codecs, so mask them out
      */
      if (capability & p->prefcodec) {
         format_t codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;

         add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
         alreadysent |= codec;
      }

      /* Start by sending our preferred audio/video codecs */
      for (x = 0; x < 64; x++) {
         format_t codec;

         if (!(codec = ast_codec_pref_index(&p->prefs, x)))
            break;

         if (!(capability & codec))
            continue;

         if (alreadysent & codec)
            continue;

         add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
         alreadysent |= codec;
      }

      /* Now send any other common audio and video codecs, and non-codec formats: */
      for (x = 1ULL; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
         if (!(capability & x))  /* Codec not requested */
            continue;

         if (alreadysent & x) /* Already added to SDP */
            continue;

         if (x & AST_FORMAT_AUDIO_MASK)
            add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
         else if (x & AST_FORMAT_VIDEO_MASK)
            add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
         else if (x & AST_FORMAT_TEXT_MASK)
            add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
      }

      /* Now add DTMF RFC2833 telephony-event as a codec */
      for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
         if (!(p->jointnoncodeccapability & x))
            continue;

         add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
      }

      ast_debug(3, "-- Done with adding codecs to SDP\n");

      if (!p->owner || !ast_internal_timing_enabled(p->owner))
         ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");

      if (min_audio_packet_size)
         ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);

      /* XXX don't think you can have ptime for video */
      if (min_video_packet_size)
         ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);

      /* XXX don't think you can have ptime for text */
      if (min_text_packet_size)
         ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);

      if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
          m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
          a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
         ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
   }

   if (add_t38) {
      /* Our T.38 end is */
      ast_udptl_get_us(p->udptl, &udptladdr);

      /* Determine T.38 UDPTL destination */
      if (!ast_sockaddr_isnull(&p->udptlredirip)) {
         ast_sockaddr_copy(&udptldest, &p->udptlredirip);
      } else {
         ast_sockaddr_copy(&udptldest, &p->ourip);
         ast_sockaddr_set_port(&udptldest, ast_sockaddr_port(&udptladdr));
      }

      if (debug) {
         ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_sockaddr_stringify_addr(&p->ourip), ast_sockaddr_port(&udptladdr));
      }

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ast_sockaddr_port(&udptldest));

      if (!ast_sockaddr_cmp(&udptldest, &dest)) {
         ast_str_append(&m_modem, 0, "c=IN %s %s\r\n",
               (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
               "IP6" : "IP4", ast_sockaddr_stringify_addr(&udptldest));
      }

      ast_str_append(&a_modem, 0, "a=T38FaxVersion:%d\r\n", p->t38.our_parms.version);
      ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", t38_get_rate(p->t38.our_parms.rate));
      if (p->t38.our_parms.fill_bit_removal) {
         ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
      }
      if (p->t38.our_parms.transcoding_mmr) {
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
      }
      if (p->t38.our_parms.transcoding_jbig) {
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
      }
      switch (p->t38.our_parms.rate_management) {
      case AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF:
         ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:transferredTCF\r\n");
         break;
      case AST_T38_RATE_MANAGEMENT_LOCAL_TCF:
         ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:localTCF\r\n");
         break;
      }
      ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%u\r\n", ast_udptl_get_local_max_datagram(p->udptl));
      switch (ast_udptl_get_error_correction_scheme(p->udptl)) {
      case UDPTL_ERROR_CORRECTION_NONE:
         break;
      case UDPTL_ERROR_CORRECTION_FEC:
         ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPFEC\r\n");
         break;
      case UDPTL_ERROR_CORRECTION_REDUNDANCY:
         ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPRedundancy\r\n");
         break;
      }
   }

   if (needaudio)
      ast_str_append(&m_audio, 0, "\r\n");
   if (needvideo)
      ast_str_append(&m_video, 0, "\r\n");
   if (needtext)
      ast_str_append(&m_text, 0, "\r\n");

   add_header(resp, "Content-Type", "application/sdp");
   add_content(resp, version);
   add_content(resp, owner);
   add_content(resp, subject);
   add_content(resp, connection);
   if (needvideo)    /* only if video response is appropriate */
      add_content(resp, bandwidth);
   add_content(resp, session_time);
   if (needaudio) {
      add_content(resp, m_audio->str);
      add_content(resp, a_audio->str);
      add_content(resp, hold);
      if (a_crypto) {
         add_content(resp, a_crypto);
      }
   } else if (p->offered_media[SDP_AUDIO].offered) {
      snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs);
      add_content(resp, dummy_answer);
   }
   if (needvideo) { /* only if video response is appropriate */
      add_content(resp, m_video->str);
      add_content(resp, a_video->str);
      add_content(resp, hold);   /* Repeat hold for the video stream */
      if (v_a_crypto) {
         add_content(resp, v_a_crypto);
      }
   } else if (p->offered_media[SDP_VIDEO].offered) {
      snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs);
      add_content(resp, dummy_answer);
   }
   if (needtext) { /* only if text response is appropriate */
      add_content(resp, m_text->str);
      add_content(resp, a_text->str);
      add_content(resp, hold);   /* Repeat hold for the text stream */
      if (t_a_crypto) {
         add_content(resp, t_a_crypto);
      }
   } else if (p->offered_media[SDP_TEXT].offered) {
      snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs);
      add_content(resp, dummy_answer);
   }
   if (add_t38) {
      add_content(resp, m_modem->str);
      add_content(resp, a_modem->str);
   } else if (p->offered_media[SDP_IMAGE].offered) {
      add_content(resp, "m=image 0 udptl t38\r\n");
   }

   /* Update lastrtprx when we send our SDP */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));

   return AST_SUCCESS;
}

static int add_sip_domain	(	const char *	domain,
		const enum domain_mode	mode,
		const char *	context
	)			[static]

Add SIP domain to list of domains we are responsible for.

Definition at line 25285 of file chan_sip.c.

References ast_calloc, ast_copy_string(), ast_debug, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_UNLOCK, ast_log(), ast_strlen_zero(), and LOG_WARNING.

Referenced by reload_config().

{
   struct domain *d;

   if (ast_strlen_zero(domain)) {
      ast_log(LOG_WARNING, "Zero length domain.\n");
      return 1;
   }

   if (!(d = ast_calloc(1, sizeof(*d))))
      return 0;

   ast_copy_string(d->domain, domain, sizeof(d->domain));

   if (!ast_strlen_zero(context))
      ast_copy_string(d->context, context, sizeof(d->context));

   d->mode = mode;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_INSERT_TAIL(&domain_list, d, list);
   AST_LIST_UNLOCK(&domain_list);

   if (sipdebug)  
      ast_debug(1, "Added local SIP domain '%s'\n", domain);

   return 1;
}

static int add_supported_header	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)			[static]

Add "Supported" header to sip message. Since some options may be disabled in the config, the sip_pvt must be inspected to determine what is supported for this dialog.

Definition at line 9020 of file chan_sip.c.

References add_header(), and st_get_mode().

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_refer(), transmit_reinvite_with_sdp(), and update_connectedline().

{
   int res;
   if (st_get_mode(pvt) != SESSION_TIMER_MODE_REFUSE) {
      res = add_header(req, "Supported", "replaces, timer");
   } else {
      res = add_header(req, "Supported", "replaces");
   }
   return res;
}

static void add_tcodec_to_sdp	(	const struct sip_pvt *	p,
		int	codec,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add text codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 10213 of file chan_sip.c.

References AST_FORMAT_T140, AST_FORMAT_T140RED, ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), and ast_verbose.

Referenced by add_sdp().

{
   int rtp_code;

   if (!p->trtp)
      return;

   if (debug)
      ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));

   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(1, codec, 0),
             ast_rtp_lookup_sample_rate2(1, codec));
   /* Add fmtp code here */

   if (codec == AST_FORMAT_T140RED) {
      int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140);
      ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
          t140code,
          t140code,
          t140code);

   }
}

static int add_text	(	struct sip_request *	req,
		const char *	text
	)			[static]

Add text body to SIP message.

Definition at line 9981 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_message_with_text().

{
   /* XXX Convert \n's to \r\n's XXX */
   add_header(req, "Content-Type", "text/plain;charset=UTF-8");
   add_content(req, text);
   return 0;
}

static struct ast_variable* add_var	(	const char *	buf,
		struct ast_variable *	list
	)			[static, read]

implement the setvar config line

Definition at line 25434 of file chan_sip.c.

References ast_strdupa, ast_variable_new(), and ast_variable::next.

{
   struct ast_variable *tmpvar = NULL;
   char *varname = ast_strdupa(buf), *varval = NULL;
   
   if ((varval = strchr(varname, '='))) {
      *varval++ = '\0';
      if ((tmpvar = ast_variable_new(varname, varval, ""))) {
         tmpvar->next = list;
         list = tmpvar;
      }
   }
   return list;
}

static void add_vcodec_to_sdp	(	const struct sip_pvt *	p,
		format_t	codec,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)			[static]

Add video codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 10190 of file chan_sip.c.

References ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), and ast_verbose.

Referenced by add_sdp().

{
   int rtp_code;

   if (!p->vrtp)
      return;

   if (debug)
      ast_verbose("Adding video codec 0x%" PRIx64 " (%s) to SDP\n", codec, ast_getformatname(codec));

   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(1, codec, 0),
             ast_rtp_lookup_sample_rate2(1, codec));
   /* Add fmtp code here */
}

static int add_vidupdate

(

struct sip_request *

req

)

[static]

add XML encoded media control with update

Note:

XML: The only way to turn 0 bits of information into a few hundred. (markster)

Definition at line 10111 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_info_with_vidupdate().

{
   const char *xml_is_a_huge_waste_of_space =
      "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
      " <media_control>\r\n"
      "  <vc_primitive>\r\n"
      "   <to_encoder>\r\n"
      "    <picture_fast_update>\r\n"
      "    </picture_fast_update>\r\n"
      "   </to_encoder>\r\n"
      "  </vc_primitive>\r\n"
      " </media_control>\r\n";
   add_header(req, "Content-Type", "application/media_control+xml");
   add_content(req, xml_is_a_huge_waste_of_space);
   return 0;
}

static void append_date

(

struct sip_request *

req

)

[static]

Append date to SIP message.

Definition at line 9819 of file chan_sip.c.

References add_header().

{
   char tmpdat[256];
   struct tm tm;
   time_t t = time(NULL);

   gmtime_r(&t, &tm);
   strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm);
   add_header(req, "Date", tmpdat);
}

static void append_history_full	(	struct sip_pvt *	p,
		const char *	fmt,
			...
	)			[static]

Append to SIP dialog history with arg list.

Definition at line 3272 of file chan_sip.c.

References append_history_va().

{
   va_list ap;

   if (!p) {
      return;
   }

   if (!p->do_history && !recordhistory && !dumphistory) {
      return;
   }

   va_start(ap, fmt);
   append_history_va(p, fmt, ap);
   va_end(ap);

   return;
}

static void append_history_va	(	struct sip_pvt *	p,
		const char *	fmt,
		va_list	ap
	)			[static]

Append to SIP dialog history with arg list.

Definition at line 3244 of file chan_sip.c.

References ast_calloc, ast_free, AST_LIST_INSERT_TAIL, AST_LIST_REMOVE_HEAD, and strsep().

Referenced by append_history_full().

{
   char buf[80], *c = buf; /* max history length */
   struct sip_history *hist;
   int l;

   vsnprintf(buf, sizeof(buf), fmt, ap);
   strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
   l = strlen(buf) + 1;
   if (!(hist = ast_calloc(1, sizeof(*hist) + l))) {
      return;
   }
   if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
      ast_free(hist);
      return;
   }
   memcpy(hist->event, buf, l);
   if (p->history_entries == MAX_HISTORY_ENTRIES) {
      struct sip_history *oldest;
      oldest = AST_LIST_REMOVE_HEAD(p->history, list);
      p->history_entries--;
      ast_free(oldest);
   }
   AST_LIST_INSERT_TAIL(p->history, hist, list);
   p->history_entries++;
}

static int apply_directmedia_ha	(	struct sip_pvt *	p,
		const char *	op
	)			[static]

Definition at line 27293 of file chan_sip.c.

References ast_apply_ha(), ast_debug, ast_rtp_instance_get_local_address(), ast_rtp_instance_get_remote_address(), AST_SENSE_ALLOW, AST_SENSE_DENY, ast_sockaddr_stringify(), and ast_strdupa.

Referenced by sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), and sip_get_vrtp_peer().

{
   struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
   int res = AST_SENSE_ALLOW;

   ast_rtp_instance_get_remote_address(p->rtp, &them);
   ast_rtp_instance_get_local_address(p->rtp, &us);

   if ((res = ast_apply_ha(p->directmediaha, &them)) == AST_SENSE_DENY) {
      ast_debug(3, "Reinvite %s to %s denied by directmedia ACL on %s\n",
         op, ast_strdupa(ast_sockaddr_stringify(&them)), ast_strdupa(ast_sockaddr_stringify(&us)));
   }

   return res;
}

AST_DATA_STRUCTURE	(	sip_peer	,
		DATA_EXPORT_SIP_PEER
	)

static void ast_quiet_chan

(

struct ast_channel *

chan

)

[static]

Turn off generator data XXX Does this function belong in the SIP channel?

Definition at line 20166 of file chan_sip.c.

References ast_channel::_state, ast_deactivate_generator(), AST_FLAG_MOH, ast_moh_stop(), AST_STATE_UP, ast_test_flag, and ast_channel::generatordata.

Referenced by attempt_transfer(), and handle_invite_replaces().

{
   if (chan && chan->_state == AST_STATE_UP) {
      if (ast_test_flag(chan, AST_FLAG_MOH))
         ast_moh_stop(chan);
      else if (chan->generatordata)
         ast_deactivate_generator(chan);
   }
}

static void ast_sip_ouraddrfor	(	const struct ast_sockaddr *	them,
		struct ast_sockaddr *	us,
		struct sip_pvt *	p
	)			[static]

NAT fix - decide which IP address to use for Asterisk server?

Using the localaddr structure built up with localnet statements in sip.conf apply it to their address to see if we need to substitute our externaddr or can get away with our internal bindaddr 'us' is always overwritten.

Definition at line 3138 of file chan_sip.c.

References ast_apply_ha(), ast_debug, ast_log(), ast_ouraddrfor(), AST_SENSE_ALLOW, ast_sockaddr_copy(), ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_stringify(), bindaddr, externaddr, get_transport(), internip, ast_tcptls_session_args::local_address, LOG_NOTICE, LOG_WARNING, and sip_cfg.

Referenced by __sip_subscribe_mwi_do(), sip_alloc(), sip_cc_monitor_request_cc(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_publish(), transmit_register(), and transmit_response_using_temp().

{
   struct ast_sockaddr theirs;

   /* Set want_remap to non-zero if we want to remap 'us' to an externally
    * reachable IP address and port. This is done if:
    * 1. we have a localaddr list (containing 'internal' addresses marked
    *    as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
    *    and AST_SENSE_ALLOW on 'external' ones);
    * 2. externaddr is set, so we know what to use as the
    *    externally visible address;
    * 3. the remote address, 'them', is external;
    * 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
    *    when passed to ast_apply_ha() so it does need to be remapped.
    *    This fourth condition is checked later.
    */
   int want_remap = 0;

   ast_sockaddr_copy(us, &internip); /* starting guess for the internal address */
   /* now ask the system what would it use to talk to 'them' */
   ast_ouraddrfor(them, us);
   ast_sockaddr_copy(&theirs, them);

   if (ast_sockaddr_is_ipv6(&theirs)) {
      if (localaddr && !ast_sockaddr_isnull(&externaddr)) {
         ast_log(LOG_WARNING, "Address remapping activated in sip.conf "
            "but we're using IPv6, which doesn't need it. Please "
            "remove \"localnet\" and/or \"externaddr\" settings.\n");
      }
   } else {
      want_remap = localaddr &&
         !ast_sockaddr_isnull(&externaddr) &&
         ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;
   }

   if (want_remap &&
       (!sip_cfg.matchexternaddrlocally || !ast_apply_ha(localaddr, us)) ) {
      /* if we used externhost, see if it is time to refresh the info */
      if (externexpire && time(NULL) >= externexpire) {
         if (ast_sockaddr_resolve_first(&externaddr, externhost, 0)) {
            ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
         }
         externexpire = time(NULL) + externrefresh;
      }
      if (!ast_sockaddr_isnull(&externaddr)) {
         ast_sockaddr_copy(us, &externaddr);
         switch (p->socket.type) {
         case SIP_TRANSPORT_TCP:
            if (!externtcpport && ast_sockaddr_port(&externaddr)) {
               /* for consistency, default to the externaddr port */
               externtcpport = ast_sockaddr_port(&externaddr);
            }
            ast_sockaddr_set_port(us, externtcpport);
            break;
         case SIP_TRANSPORT_TLS:
            ast_sockaddr_set_port(us, externtlsport);
            break;
         case SIP_TRANSPORT_UDP:
            if (!ast_sockaddr_port(&externaddr)) {
               ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
            }
            break;
         default:
            break;
         }
      }
      ast_debug(1, "Target address %s is not local, substituting externaddr\n",
           ast_sockaddr_stringify(them));
   } else if (p) {
      /* no remapping, but we bind to a specific address, so use it. */
      switch (p->socket.type) {
      case SIP_TRANSPORT_TCP:
         if (!ast_sockaddr_is_any(&sip_tcp_desc.local_address)) {
            ast_sockaddr_copy(us,
                    &sip_tcp_desc.local_address);
         } else {
            ast_sockaddr_set_port(us,
                        ast_sockaddr_port(&sip_tcp_desc.local_address));
         }
         break;
      case SIP_TRANSPORT_TLS:
         if (!ast_sockaddr_is_any(&sip_tls_desc.local_address)) {
            ast_sockaddr_copy(us,
                    &sip_tls_desc.local_address);
         } else {
            ast_sockaddr_set_port(us,
                        ast_sockaddr_port(&sip_tls_desc.local_address));
         }
         break;
      case SIP_TRANSPORT_UDP:
         /* fall through on purpose */
      default:
         if (!ast_sockaddr_is_any(&bindaddr)) {
            ast_sockaddr_copy(us, &bindaddr);
         }
         if (!ast_sockaddr_port(us)) {
            ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
         }
      }
   } else if (!ast_sockaddr_is_any(&bindaddr)) {
      ast_sockaddr_copy(us, &bindaddr);
   }
   ast_debug(3, "Setting SIP_TRANSPORT_%s with address %s\n", get_transport(p->socket.type), ast_sockaddr_stringify(us));
}

static int ast_sockaddr_resolve_first	(	struct ast_sockaddr *	addr,
		const char *	name,
		int	flag
	)			[static]

Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.

Using this function probably means you have a faulty design.

Definition at line 27955 of file chan_sip.c.

References ast_sockaddr_resolve_first_af(), bindaddr, and get_address_family_filter().

Referenced by __set_address_from_contact(), ast_sip_ouraddrfor(), check_via(), create_addr(), parse_register_contact(), reload_config(), set_destination(), sip_do_debug_ip(), and sip_request_call().

{
   return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(&bindaddr));
}

static int ast_sockaddr_resolve_first_af	(	struct ast_sockaddr *	addr,
		const char *	name,
		int	flag,
		int	family
	)			[static]

Return the first entry from ast_sockaddr_resolve filtered by address family.

Using this function probably means you have a faulty design.

Definition at line 27931 of file chan_sip.c.

References ast_debug, ast_free, ast_sockaddr_copy(), and ast_sockaddr_resolve().

Referenced by ast_sockaddr_resolve_first(), get_ip_and_port_from_sdp(), and process_sdp_c().

{
   struct ast_sockaddr *addrs;
   int addrs_cnt;

   addrs_cnt = ast_sockaddr_resolve(&addrs, name, flag, family);
   if (addrs_cnt <= 0) {
      return 1;
   }
   if (addrs_cnt > 1) {
      ast_debug(1, "Multiple addresses, using the first one only\n");
   }

   ast_sockaddr_copy(addr, &addrs[0]);

   ast_free(addrs);
   return 0;
}

static int attempt_transfer	(	struct sip_dual *	transferer,
		struct sip_dual *	target
	)			[static]

Attempt transfer of SIP call This fix for attended transfers on a local PBX.

Definition at line 20178 of file chan_sip.c.

References ast_channel_masquerade(), ast_debug, ast_log(), ast_quiet_chan(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_state2str(), LOG_NOTICE, and LOG_WARNING.

{
   int res = 0;
   struct ast_channel *peera = NULL,   
      *peerb = NULL,
      *peerc = NULL,
      *peerd = NULL;


   /* We will try to connect the transferee with the target and hangup
      all channels to the transferer */   
   ast_debug(4, "Sip transfer:--------------------\n");
   if (transferer->chan1)
      ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state));
   else
      ast_debug(4, "-- No transferer first channel - odd??? \n");
   if (target->chan1)
      ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state));
   else
      ast_debug(4, "-- No target first channel ---\n");
   if (transferer->chan2)
      ast_debug(4, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state));
   else
      ast_debug(4, "-- No bridged call to transferee\n");
   if (target->chan2)
      ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)");
   else
      ast_debug(4, "-- No target second channel ---\n");
   ast_debug(4, "-- END Sip transfer:--------------------\n");
   if (transferer->chan2) { /* We have a bridge on the transferer's channel */
      peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
      peerb = target->chan1;     /* Transferer - PBX -> target channel - This will get lost in masq */
      peerc = transferer->chan2; /* Asterisk to Transferee */
      peerd = target->chan2;     /* Asterisk to Target */
      ast_debug(3, "SIP transfer: Four channels to handle\n");
   } else if (target->chan2) {   /* Transferer has no bridge (IVR), but transferee */
      peera = target->chan1;     /* Transferer to PBX -> target channel */
      peerb = transferer->chan1; /* Transferer to IVR*/
      peerc = target->chan2;     /* Asterisk to Target */
      peerd = transferer->chan2; /* Nothing */
      ast_debug(3, "SIP transfer: Three channels to handle\n");
   }

   if (peera && peerb && peerc && (peerb != peerc)) {
      ast_quiet_chan(peera);     /* Stop generators */
      ast_quiet_chan(peerb);  
      ast_quiet_chan(peerc);
      if (peerd)
         ast_quiet_chan(peerd);

      ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name);
      if (ast_channel_masquerade(peerb, peerc)) {
         ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name);
         res = -1;
      } else
         ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n");
      return res;
   } else {
      ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n");
      if (transferer->chan1)
         ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
      if (target->chan1)
         ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
      return -1;
   }
   return 0;
}

static void auth_headers	(	enum sip_auth_type	code,
		char **	header,
		char **	respheader
	)			[static]

return the request and response heade for a 401 or 407 code

Definition at line 12612 of file chan_sip.c.

References ast_verbose.

Referenced by check_auth(), do_proxy_auth(), do_register_auth(), and transmit_request_with_auth().

{
   if (code == WWW_AUTH) {       /* 401 */
      *header = "WWW-Authenticate";
      *respheader = "Authorization";
   } else if (code == PROXY_AUTH) { /* 407 */
      *header = "Proxy-Authenticate";
      *respheader = "Proxy-Authorization";
   } else {
      ast_verbose("-- wrong response code %d\n", code);
      *header = *respheader = "Invalid";
   }
}

static int auto_congest

(

const void *

arg

)

[static]

Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.

Definition at line 5049 of file chan_sip.c.

References append_history, ast_channel_trylock, ast_channel_unlock, AST_CONTROL_CONGESTION, ast_queue_control(), sip_pvt_lock, sip_pvt_unlock, and sip_scheddestroy().

{
   struct sip_pvt *p = (struct sip_pvt *)arg;

   sip_pvt_lock(p);
   p->initid = -1;   /* event gone, will not be rescheduled */
   if (p->owner) {
      /* XXX fails on possible deadlock */
      if (!ast_channel_trylock(p->owner)) {
         append_history(p, "Cong", "Auto-congesting (timer)");
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         ast_channel_unlock(p->owner);
      }

      /* Give the channel a chance to act before we proceed with destruction */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }
   sip_pvt_unlock(p);
   dialog_unref(p, "unreffing arg passed into auto_congest callback (p->initid)");
   return 0;
}

static void build_callid_pvt

(

struct sip_pvt *

pvt

)

[static]

Build SIP Call-ID value for a non-REGISTER transaction.

Definition at line 6983 of file chan_sip.c.

References ast_sockaddr_stringify(), ast_string_field_build, generate_random_string(), and S_OR.

Referenced by __sip_subscribe_mwi_do(), sip_alloc(), sip_poke_peer(), sip_request_call(), and sip_send_mwi_to_peer().

{
   char buf[33];

   const char *host = S_OR(pvt->fromdomain, ast_sockaddr_stringify(&pvt->ourip));
   
   ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);

}

static void build_callid_registry	(	struct sip_registry *	reg,
		const struct ast_sockaddr *	ourip,
		const char *	fromdomain
	)			[static]

Build SIP Call-ID value for a REGISTER transaction.

Definition at line 6994 of file chan_sip.c.

References ast_sockaddr_stringify_host(), ast_string_field_build, generate_random_string(), and S_OR.

Referenced by transmit_register().

{
   char buf[33];

   const char *host = S_OR(fromdomain, ast_sockaddr_stringify_host(ourip));

   ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}

static void build_contact

(

struct sip_pvt *

p

)

[static]

Build contact header - the contact header we send out.

Definition at line 11013 of file chan_sip.c.

References ast_sockaddr_stringify(), ast_string_field_build, ast_strlen_zero(), and get_transport().

Referenced by __sip_subscribe_mwi_do(), check_user_full(), handle_request_invite(), handle_request_options(), handle_request_subscribe(), initreqprep(), register_verify(), and transmit_register().

{
   if (p->socket.type == SIP_TRANSPORT_UDP) {
      ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten,
         ast_strlen_zero(p->exten) ? "" : "@", ast_sockaddr_stringify(&p->ourip));
   } else {
      ast_string_field_build(p, our_contact, "<sip:%s%s%s;transport=%s>", p->exten,
         ast_strlen_zero(p->exten) ? "" : "@", ast_sockaddr_stringify(&p->ourip),
         get_transport(p->socket.type));
   }
}

static struct sip_peer * build_peer	(	const char *	name,
		struct ast_variable *	v,
		struct ast_variable *	alt,
		int	realtime,
		int	devstate_only
	)			[static, read]

Build peer from configuration (file or realtime static/dynamic).

< The first transport listed should be default outbound

Definition at line 25579 of file chan_sip.c.

References __set_address_from_contact(), accountcode, add_peer_mailboxes(), add_peer_mwi_subs(), add_realm_authentication(), add_var(), ao2_alloc, ao2_t_alloc, ao2_t_find, ao2_t_ref, asprintf, ast_append_ha(), ast_atomic_fetchadd_int(), ast_callerid_split(), AST_CC_AGENT_NATIVE, AST_CC_AGENT_NEVER, ast_cc_config_params_init, ast_cc_is_config_param(), ast_cc_set_param(), ast_cdr_amaflags2int(), ast_copy_flags, ast_copy_string(), ast_debug, ast_dnsmgr_lookup(), ast_free, ast_free_ha(), ast_get_cc_agent_policy(), ast_get_group(), ast_get_ip(), ast_get_time_t(), AST_LIST_EMPTY, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, ast_log(), ast_parse_allow_disallow(), ast_parse_caller_presentation(), AST_SCHED_DEL_UNREF, ast_set2_flag, ast_set_cc_agent_policy(), ast_set_flag, ast_skip_blanks(), ast_sockaddr_isnull(), ast_sockaddr_parse(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_setnull(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_host(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_reset(), ast_str_set(), ast_str_strlen(), ast_strdupa, ast_string_field_init, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_true(), ast_variables_destroy(), bindaddr, cid_name, cid_num, clear_realm_authentication(), context, DEFAULT_MAXMS, destroy_association(), destroy_mailbox(), errno, FALSE, ast_flags::flags, format, get_address_family_filter(), get_srv_protocol(), get_srv_service(), handle_common_options(), handle_t38_options(), language, ast_variable::lineno, LOG_ERROR, LOG_NOTICE, LOG_WARNING, mailbox, mark_parsed_methods(), MAXHOSTNAMELEN, mohinterpret, mohsuggest, ast_variable::name, ast_variable::next, OBJ_POINTER, OBJ_UNLINK, parkinglot, PARSE_PORT_FORBID, port_str2int(), proxy_update(), ref_peer(), reg_source_db(), secret, set_peer_defaults(), set_socket_transport(), sip_cfg, sip_destroy_peer_fn(), sip_poke_peer(), sip_register(), sip_send_mwi_to_peer(), srvlookup, str2stmode(), str2strefresher(), strsep(), TRUE, unref_peer(), and ast_variable::value.

{
   struct sip_peer *peer = NULL;
   struct ast_ha *oldha = NULL;
   struct ast_ha *olddirectmediaha = NULL;
   int found = 0;
   int firstpass = 1;
   uint16_t port = 0;
   int format = 0;      /* Ama flags */
   int timerb_set = 0, timert1_set = 0;
   time_t regseconds = 0;
   struct ast_flags peerflags[3] = {{(0)}};
   struct ast_flags mask[3] = {{(0)}};
   char callback[256] = "";
   struct sip_peer tmp_peer;
   const char *srvlookup = NULL;
   static int deprecation_warning = 1;
   int alt_fullcontact = alt ? 1 : 0, headercount = 0;
   struct ast_str *fullcontact = ast_str_alloca(512);

   if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      /* Note we do NOT use find_peer here, to avoid realtime recursion */
      /* We also use a case-sensitive comparison (unlike find_peer) so
         that case changes made to the peer name will be properly handled
         during reload
      */
      ast_copy_string(tmp_peer.name, name, sizeof(tmp_peer.name));
      peer = ao2_t_find(peers, &tmp_peer, OBJ_POINTER | OBJ_UNLINK, "find and unlink peer from peers table");
   }

   if (peer) {
      /* Already in the list, remove it and it will be added back (or FREE'd)  */
      found++;
      if (!(peer->the_mark))
         firstpass = 0;
   } else {
      if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
         return NULL;

      if (ast_string_field_init(peer, 512)) {
         ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
         return NULL;
      }

      if (!(peer->cc_params = ast_cc_config_params_init())) {
         ao2_t_ref(peer, -1, "failed to allocate cc_params for peer");
         return NULL;
      }

      if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
         ast_atomic_fetchadd_int(&rpeerobjs, 1);
         ast_debug(3, "-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
      } else
         ast_atomic_fetchadd_int(&speerobjs, 1);
   }

   /* Note that our peer HAS had its reference count increased */
   if (firstpass) {
      peer->lastmsgssent = -1;
      oldha = peer->ha;
      peer->ha = NULL;
      olddirectmediaha = peer->directmediaha;
      peer->directmediaha = NULL;
      set_peer_defaults(peer);   /* Set peer defaults */
      peer->type = 0;
   }
   if (!found && name)
      ast_copy_string(peer->name, name, sizeof(peer->name));

   /* If we have channel variables, remove them (reload) */
   if (peer->chanvars) {
      ast_variables_destroy(peer->chanvars);
      peer->chanvars = NULL;
      /* XXX should unregister ? */
   }

   if (found)
      peer->portinuri = 0;

   /* If we have realm authentication information, remove them (reload) */
   clear_realm_authentication(peer->auth);
   peer->auth = NULL;
   /* clear the transport information.  We will detect if a default value is required after parsing the config */
   peer->default_outbound_transport = 0;
   peer->transports = 0;

   if (!devstate_only) {
      struct sip_mailbox *mailbox;
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         mailbox->delme = 1;
      }
   }

   for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
      if (!devstate_only) {
         if (handle_common_options(&peerflags[0], &mask[0], v)) {
            continue;
         }
         if (handle_t38_options(&peerflags[0], &mask[0], v, &peer->t38_maxdatagram)) {
            continue;
         }
         if (!strcasecmp(v->name, "transport") && !ast_strlen_zero(v->value)) {
            char *val = ast_strdupa(v->value);
            char *trans;

            while ((trans = strsep(&val, ","))) {
               trans = ast_skip_blanks(trans);

               if (!strncasecmp(trans, "udp", 3)) {
                  peer->transports |= SIP_TRANSPORT_UDP;
               } else if (!strncasecmp(trans, "tcp", 3)) {
                  peer->transports |= SIP_TRANSPORT_TCP;
               } else if (!strncasecmp(trans, "tls", 3)) {
                  peer->transports |= SIP_TRANSPORT_TLS;
               } else {
                  ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
               }

               if (!peer->default_outbound_transport) { /*!< The first transport listed should be default outbound */
                  peer->default_outbound_transport = peer->transports;
               }
            }
         } else if (realtime && !strcasecmp(v->name, "regseconds")) {
            ast_get_time_t(v->value, &regseconds, 0, NULL);
         } else if (realtime && !strcasecmp(v->name, "name")) {
            ast_copy_string(peer->name, v->value, sizeof(peer->name));
         } else if (realtime && !strcasecmp(v->name, "useragent")) {
            ast_string_field_set(peer, useragent, v->value);
         } else if (!strcasecmp(v->name, "type")) {
            if (!strcasecmp(v->value, "peer")) {
               peer->type |= SIP_TYPE_PEER;
            } else if (!strcasecmp(v->value, "user")) {
               peer->type |= SIP_TYPE_USER;
            } else if (!strcasecmp(v->value, "friend")) {
               peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
            }
         } else if (!strcasecmp(v->name, "remotesecret")) {
            ast_string_field_set(peer, remotesecret, v->value);
         } else if (!strcasecmp(v->name, "secret")) {
            ast_string_field_set(peer, secret, v->value);
         } else if (!strcasecmp(v->name, "md5secret")) {
            ast_string_field_set(peer, md5secret, v->value);
         } else if (!strcasecmp(v->name, "auth")) {
            peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno);
         } else if (!strcasecmp(v->name, "callerid")) {
            char cid_name[80] = { '\0' }, cid_num[80] = { '\0' };

            ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
            ast_string_field_set(peer, cid_name, cid_name);
            ast_string_field_set(peer, cid_num, cid_num);
         } else if (!strcasecmp(v->name, "mwi_from")) {
            ast_string_field_set(peer, mwi_from, v->value);
         } else if (!strcasecmp(v->name, "fullname")) {
            ast_string_field_set(peer, cid_name, v->value);
         } else if (!strcasecmp(v->name, "trunkname")) {
            /* This is actually for a trunk, so we don't want to override callerid */
            ast_string_field_set(peer, cid_name, "");
         } else if (!strcasecmp(v->name, "cid_number")) {
            ast_string_field_set(peer, cid_num, v->value);
         } else if (!strcasecmp(v->name, "cid_tag")) {
            ast_string_field_set(peer, cid_tag, v->value);
         } else if (!strcasecmp(v->name, "context")) {
            ast_string_field_set(peer, context, v->value);
            ast_set_flag(&peer->flags[1], SIP_PAGE2_HAVEPEERCONTEXT);
         } else if (!strcasecmp(v->name, "subscribecontext")) {
            ast_string_field_set(peer, subscribecontext, v->value);
         } else if (!strcasecmp(v->name, "fromdomain")) {
            char *fromdomainport;
            ast_string_field_set(peer, fromdomain, v->value);
            if ((fromdomainport = strchr(peer->fromdomain, ':'))) {
               *fromdomainport++ = '\0';
               if (!(peer->fromdomainport = port_str2int(fromdomainport, 0))) {
                  ast_log(LOG_NOTICE, "'%s' is not a valid port number for fromdomain.\n",fromdomainport);
               }
            } else {
               peer->fromdomainport = STANDARD_SIP_PORT;
            }
         } else if (!strcasecmp(v->name, "usereqphone")) {
            ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);
         } else if (!strcasecmp(v->name, "fromuser")) {
            ast_string_field_set(peer, fromuser, v->value);
         } else if (!strcasecmp(v->name, "outboundproxy")) {
            char *tok, *proxyname;

            if (ast_strlen_zero(v->value)) {
               ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf.", v->lineno);
               continue;
            }

            peer->outboundproxy =
                ao2_alloc(sizeof(*peer->outboundproxy), NULL);

            tok = ast_skip_blanks(strtok(ast_strdupa(v->value), ","));

            sip_parse_host(tok, v->lineno, &proxyname,
                      &peer->outboundproxy->port,
                      &peer->outboundproxy->transport);

            tok = ast_skip_blanks(strtok(ast_strdupa(v->value), ","));

            if ((tok = strtok(NULL, ","))) {
               peer->outboundproxy->force = !strncasecmp(ast_skip_blanks(tok), "force", 5);
            } else {
               peer->outboundproxy->force = FALSE;
            }

            if (ast_strlen_zero(proxyname)) {
               ast_log(LOG_WARNING, "you must specify a name for the outboundproxy on line %d of sip.conf.", v->lineno);
               sip_cfg.outboundproxy.name[0] = '\0';
               continue;
            }

            ast_copy_string(peer->outboundproxy->name, proxyname, sizeof(peer->outboundproxy->name));

            proxy_update(peer->outboundproxy);
         } else if (!strcasecmp(v->name, "host")) {
            if (!strcasecmp(v->value, "dynamic")) {
               /* They'll register with us */
               if (!found || !peer->host_dynamic) {
                  /* Initialize stuff if this is a new peer, or if it used to
                   * not be dynamic before the reload. */
                  ast_sockaddr_setnull(&peer->addr);
               }
               peer->host_dynamic = TRUE;
            } else {
               /* Non-dynamic.  Make sure we become that way if we're not */
               AST_SCHED_DEL_UNREF(sched, peer->expire,
                     unref_peer(peer, "removing register expire ref"));
               peer->host_dynamic = FALSE;
               srvlookup = v->value;
            }
         } else if (!strcasecmp(v->name, "defaultip")) {
            if (!ast_strlen_zero(v->value) && ast_get_ip(&peer->defaddr, v->value)) {
               unref_peer(peer, "unref_peer: from build_peer defaultip");
               return NULL;
            }
         } else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny")) {
            int ha_error = 0;
            if (!ast_strlen_zero(v->value)) {
               peer->ha = ast_append_ha(v->name, v->value, peer->ha, &ha_error);
            }
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
            int ha_error = 0;
            if (!ast_strlen_zero(v->value)) {
               peer->contactha = ast_append_ha(v->name + 7, v->value, peer->contactha, &ha_error);
            }
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "directmediapermit") || !strcasecmp(v->name, "directmediadeny")) {
            int ha_error = 0;
            peer->directmediaha = ast_append_ha(v->name + 11, v->value, peer->directmediaha, &ha_error);
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad directmedia ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "port")) {
            peer->portinuri = 1;
            if (!(port = port_str2int(v->value, 0))) {
               if (realtime) {
                  /* If stored as integer, could be 0 for some DBs (notably MySQL) */
                  peer->portinuri = 0;
               } else {
                  ast_log(LOG_WARNING, "Invalid peer port configuration at line %d : %s\n", v->lineno, v->value);
               }
            }
         } else if (!strcasecmp(v->name, "callingpres")) {
            peer->callingpres = ast_parse_caller_presentation(v->value);
            if (peer->callingpres == -1) {
               peer->callingpres = atoi(v->value);
            }
         } else if (!strcasecmp(v->name, "username") || !strcmp(v->name, "defaultuser")) {   /* "username" is deprecated */
            ast_string_field_set(peer, username, v->value);
            if (!strcasecmp(v->name, "username")) {
               if (deprecation_warning) {
                  ast_log(LOG_NOTICE, "The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'\n");
                  deprecation_warning = 0;
               }
               peer->deprecated_username = 1;
            }
         } else if (!strcasecmp(v->name, "language")) {
            ast_string_field_set(peer, language, v->value);
         } else if (!strcasecmp(v->name, "regexten")) {
            ast_string_field_set(peer, regexten, v->value);
         } else if (!strcasecmp(v->name, "callbackextension")) {
            ast_copy_string(callback, v->value, sizeof(callback));
         } else if (!strcasecmp(v->name, "amaflags")) {
            format = ast_cdr_amaflags2int(v->value);
            if (format < 0) {
               ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
            } else {
               peer->amaflags = format;
            }
         } else if (!strcasecmp(v->name, "maxforwards")) {
            if ((sscanf(v->value, "%30d", &peer->maxforwards) != 1) || (peer->maxforwards < 1)) {
               ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d.  Using default.\n", v->value, v->lineno);
               peer->maxforwards = sip_cfg.default_max_forwards;
            }
         } else if (!strcasecmp(v->name, "accountcode")) {
            ast_string_field_set(peer, accountcode, v->value);
         } else if (!strcasecmp(v->name, "mohinterpret")) {
            ast_string_field_set(peer, mohinterpret, v->value);
         } else if (!strcasecmp(v->name, "mohsuggest")) {
            ast_string_field_set(peer, mohsuggest, v->value);
         } else if (!strcasecmp(v->name, "parkinglot")) {
            ast_string_field_set(peer, parkinglot, v->value);
         } else if (!strcasecmp(v->name, "rtp_engine")) {
            ast_string_field_set(peer, engine, v->value);
         } else if (!strcasecmp(v->name, "mailbox")) {
            add_peer_mailboxes(peer, v->value);
         } else if (!strcasecmp(v->name, "hasvoicemail")) {
            /* People expect that if 'hasvoicemail' is set, that the mailbox will
             * be also set, even if not explicitly specified. */
            if (ast_true(v->value) && AST_LIST_EMPTY(&peer->mailboxes)) {
               add_peer_mailboxes(peer, name);
            }
         } else if (!strcasecmp(v->name, "subscribemwi")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
         } else if (!strcasecmp(v->name, "vmexten")) {
            ast_string_field_set(peer, vmexten, v->value);
         } else if (!strcasecmp(v->name, "callgroup")) {
            peer->callgroup = ast_get_group(v->value);
         } else if (!strcasecmp(v->name, "allowtransfer")) {
            peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
         } else if (!strcasecmp(v->name, "pickupgroup")) {
            peer->pickupgroup = ast_get_group(v->value);
         } else if (!strcasecmp(v->name, "allow")) {
            int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, TRUE);
            if (error) {
               ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
            }
         } else if (!strcasecmp(v->name, "disallow")) {
            int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, FALSE);
            if (error) {
               ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
            }
         } else if (!strcasecmp(v->name, "preferred_codec_only")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
         } else if (!strcasecmp(v->name, "registertrying")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING);
         } else if (!strcasecmp(v->name, "autoframing")) {
            peer->autoframing = ast_true(v->value);
         } else if (!strcasecmp(v->name, "rtptimeout")) {
            if ((sscanf(v->value, "%30d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtptimeout = global_rtptimeout;
            }
         } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
            if ((sscanf(v->value, "%30d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtpholdtimeout = global_rtpholdtimeout;
            }
         } else if (!strcasecmp(v->name, "rtpkeepalive")) {
            if ((sscanf(v->value, "%30d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtpkeepalive = global_rtpkeepalive;
            }
         } else if (!strcasecmp(v->name, "timert1")) {
            if ((sscanf(v->value, "%30d", &peer->timer_t1) != 1) || (peer->timer_t1 < 200) || (peer->timer_t1 < global_t1min)) {
               ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d.  Using default.\n", v->value, v->lineno);
               peer->timer_t1 = global_t1min;
            }
            timert1_set = 1;
         } else if (!strcasecmp(v->name, "timerb")) {
            if ((sscanf(v->value, "%30d", &peer->timer_b) != 1) || (peer->timer_b < 200)) {
               ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d.  Using default.\n", v->value, v->lineno);
               peer->timer_b = global_timer_b;
            }
            timerb_set = 1;
         } else if (!strcasecmp(v->name, "setvar")) {
            peer->chanvars = add_var(v->value, peer->chanvars);
         } else if (!strcasecmp(v->name, "header")) {
            char tmp[4096];
            snprintf(tmp, sizeof(tmp), "__SIPADDHEADERpre%2d=%s", ++headercount, v->value);
            peer->chanvars = add_var(tmp, peer->chanvars);
         } else if (!strcasecmp(v->name, "qualifyfreq")) {
            int i;
            if (sscanf(v->value, "%30d", &i) == 1) {
               peer->qualifyfreq = i * 1000;
            } else {
               ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->qualifyfreq = global_qualifyfreq;
            }
         } else if (!strcasecmp(v->name, "maxcallbitrate")) {
            peer->maxcallbitrate = atoi(v->value);
            if (peer->maxcallbitrate < 0) {
               peer->maxcallbitrate = default_maxcallbitrate;
            }
         } else if (!strcasecmp(v->name, "session-timers")) {
            int i = (int) str2stmode(v->value);
            if (i < 0) {
               ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_mode_oper = global_st_mode;
            } else {
               peer->stimer.st_mode_oper = i;
            }
         } else if (!strcasecmp(v->name, "session-expires")) {
            if (sscanf(v->value, "%30d", &peer->stimer.st_max_se) != 1) {
               ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_max_se = global_max_se;
            }
         } else if (!strcasecmp(v->name, "session-minse")) {
            if (sscanf(v->value, "%30d", &peer->stimer.st_min_se) != 1) {
               ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_min_se = global_min_se;
            }
            if (peer->stimer.st_min_se < 90) {
               ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < 90 secs\n", v->value, v->lineno, config);
               peer->stimer.st_min_se = global_min_se;
            }
         } else if (!strcasecmp(v->name, "session-refresher")) {
            int i = (int) str2strefresher(v->value);
            if (i < 0) {
               ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_ref = global_st_refresher;
            } else {
               peer->stimer.st_ref = i;
            }
         } else if (!strcasecmp(v->name, "disallowed_methods")) {
            char *disallow = ast_strdupa(v->value);
            mark_parsed_methods(&peer->disallowed_methods, disallow);
         } else if (!strcasecmp(v->name, "unsolicited_mailbox")) {
            ast_string_field_set(peer, unsolicited_mailbox, v->value);
         } else if (!strcasecmp(v->name, "use_q850_reason")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
         } else if (!strcasecmp(v->name, "encryption")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_USE_SRTP);
         } else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
            ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
         }
      }

      /* These apply to devstate lookups */
      if (realtime && !strcasecmp(v->name, "lastms")) {
         sscanf(v->value, "%30d", &peer->lastms);
      } else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
         ast_sockaddr_parse(&peer->addr, v->value, PARSE_PORT_FORBID);
      } else if (realtime && !strcasecmp(v->name, "fullcontact")) {
         if (alt_fullcontact && !alt) {
            /* Reset, because the alternate also has a fullcontact and we
             * do NOT want the field value to be doubled. It might be
             * tempting to skip this, but the first table might not have
             * fullcontact and since we're here, we know that the alternate
             * absolutely does. */
            alt_fullcontact = 0;
            ast_str_reset(fullcontact);
         }
         /* Reconstruct field, because realtime separates our value at the ';' */
         if (fullcontact->used > 0) {
            ast_str_append(&fullcontact, 0, ";%s", v->value);
         } else {
            ast_str_set(&fullcontact, 0, "%s", v->value);
         }
      } else if (!strcasecmp(v->name, "qualify")) {
         if (!strcasecmp(v->value, "no")) {
            peer->maxms = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
         } else if (sscanf(v->value, "%30d", &peer->maxms) != 1) {
            ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
            peer->maxms = 0;
         }
         if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
            /* This would otherwise cause a network storm, where the
             * qualify response refreshes the peer from the database,
             * which in turn causes another qualify to be sent, ad
             * infinitum. */
            ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime.  Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
            peer->maxms = 0;
         }
      } else if (!strcasecmp(v->name, "callcounter")) {
         peer->call_limit = ast_true(v->value) ? INT_MAX : 0;
      } else if (!strcasecmp(v->name, "call-limit")) {
         peer->call_limit = atoi(v->value);
         if (peer->call_limit < 0) {
            peer->call_limit = 0;
         }
      } else if (!strcasecmp(v->name, "busylevel")) {
         peer->busy_level = atoi(v->value);
         if (peer->busy_level < 0) {
            peer->busy_level = 0;
         }
      } else if (ast_cc_is_config_param(v->name)) {
         ast_cc_set_param(peer->cc_params, v->name, v->value);
      }
   }

   if (!devstate_only) {
      struct sip_mailbox *mailbox;
      AST_LIST_TRAVERSE_SAFE_BEGIN(&peer->mailboxes, mailbox, entry) {
         if (mailbox->delme) {
            AST_LIST_REMOVE_CURRENT(entry);
            destroy_mailbox(mailbox);
         }
      }
      AST_LIST_TRAVERSE_SAFE_END;
   }

   if (!can_parse_xml && (ast_get_cc_agent_policy(peer->cc_params) == AST_CC_AGENT_NATIVE)) {
      ast_log(LOG_WARNING, "Peer %s has a cc_agent_policy of 'native' but required libxml2 dependency is not installed. Changing policy to 'never'\n", peer->name);
      ast_set_cc_agent_policy(peer->cc_params, AST_CC_AGENT_NEVER);
   }

   /* Note that Timer B is dependent upon T1 and MUST NOT be lower
    * than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
   if (peer->timer_b < peer->timer_t1 * 64) {
      if (timerb_set && timert1_set) {
         ast_log(LOG_WARNING, "Timer B has been set lower than recommended for peer %s (%d < 64 * Timer-T1=%d)\n", peer->name, peer->timer_b, peer->timer_t1);
      } else if (timerb_set) {
         if ((peer->timer_t1 = peer->timer_b / 64) < global_t1min) {
            ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", peer->timer_b, peer->timer_t1);
            peer->timer_t1 = global_t1min;
            peer->timer_b = peer->timer_t1 * 64;
         }
         peer->timer_t1 = peer->timer_b / 64;
      } else {
         peer->timer_b = peer->timer_t1 * 64;
      }
   }

   if (!peer->default_outbound_transport) {
      /* Set default set of transports */
      peer->transports = default_transports;
      /* Set default primary transport */
      peer->default_outbound_transport = default_primary_transport;
   }

   /* The default transport type set during build_peer should only replace the socket.type when...
    * 1. Registration is not present and the socket.type and default transport types are different.
    * 2. The socket.type is not an acceptable transport type after rebuilding peer.
    * 3. The socket.type is not set yet. */
   if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
      !(peer->socket.type & peer->transports) || !(peer->socket.type)) {

      set_socket_transport(&peer->socket, peer->default_outbound_transport);
   }

   if (ast_str_strlen(fullcontact)) {
      ast_string_field_set(peer, fullcontact, ast_str_buffer(fullcontact));
      peer->rt_fromcontact = TRUE;
      /* We have a hostname in the fullcontact, but if we don't have an
       * address listed on the entry (or if it's 'dynamic'), then we need to
       * parse the entry to obtain the IP address, so a dynamic host can be
       * contacted immediately after reload (as opposed to waiting for it to
       * register once again). But if we have an address for this peer and NAT was
       * specified, use that address instead. */
      /* XXX May need to revisit the final argument; does the realtime DB store whether
       * the original contact was over TLS or not? XXX */
      if (!ast_test_flag(&peer->flags[0], SIP_NAT_RPORT_PRESENT) || ast_sockaddr_isnull(&peer->addr)) {
         __set_address_from_contact(fullcontact->str, &peer->addr, 0);
      }
   }

   if (srvlookup && peer->dnsmgr == NULL) {
      char transport[MAXHOSTNAMELEN];
      char _srvlookup[MAXHOSTNAMELEN];
      char *params;

      ast_copy_string(_srvlookup, srvlookup, sizeof(_srvlookup));
      if ((params = strchr(_srvlookup, ';'))) {
         *params++ = '\0';
      }

      snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(peer->socket.type), get_srv_protocol(peer->socket.type));

      peer->addr.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
      if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup && !peer->portinuri ? transport : NULL)) {
         ast_log(LOG_ERROR, "srvlookup failed for host: %s, on peer %s, removing peer\n", _srvlookup, peer->name);
         unref_peer(peer, "getting rid of a peer pointer");
         return NULL;
      }

      ast_string_field_set(peer, tohost, peer->dnsmgr ? srvlookup :
                 ast_sockaddr_stringify_host(&peer->addr));
      
      if (global_dynamic_exclude_static) {
         int err = 0;
         sip_cfg.contact_ha = ast_append_ha("deny", ast_sockaddr_stringify_addr(&peer->addr), 
                     sip_cfg.contact_ha, &err);
         if (err) {
            ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
         }
      }
   }

   if (port && !realtime && peer->host_dynamic) {
      ast_sockaddr_set_port(&peer->defaddr, port);
   } else if (port) {
      ast_sockaddr_set_port(&peer->addr, port);
   }

   if (ast_sockaddr_port(&peer->addr) == 0) {
      ast_sockaddr_set_port(&peer->addr,
                  (peer->socket.type & SIP_TRANSPORT_TLS) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }
   if (ast_sockaddr_port(&peer->defaddr) == 0) {
      ast_sockaddr_set_port(&peer->defaddr,
                  (peer->socket.type & SIP_TRANSPORT_TLS) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }
   if (!peer->socket.port) {
      peer->socket.port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));
   }

   if (!sip_cfg.ignore_regexpire && peer->host_dynamic && realtime) {
      time_t nowtime = time(NULL);

      if ((nowtime - regseconds) > 0) {
         destroy_association(peer);
         memset(&peer->addr, 0, sizeof(peer->addr));
         peer->lastms = -1;
         ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
      }
   }

   /* Startup regular pokes */
   if (!devstate_only && realtime && peer->lastms > 0) {
      ref_peer(peer, "schedule qualify");
      sip_poke_peer(peer, 0);
   }

   ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
   ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags);
   ast_copy_flags(&peer->flags[2], &peerflags[2], mask[2].flags);
   if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
      sip_cfg.allowsubscribe = TRUE;   /* No global ban any more */
   }
   if (peer->host_dynamic && !peer->is_realtime) {
      reg_source_db(peer);
   }

   /* If they didn't request that MWI is sent *only* on subscribe, go ahead and
    * subscribe to it now. */
   if (!devstate_only && !ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) &&
      !AST_LIST_EMPTY(&peer->mailboxes)) {
      add_peer_mwi_subs(peer);
      /* Send MWI from the event cache only.  This is so we can send initial
       * MWI if app_voicemail got loaded before chan_sip.  If it is the other
       * way, then we will get events when app_voicemail gets loaded. */
      sip_send_mwi_to_peer(peer, NULL, 1);
   }

   peer->the_mark = 0;

   ast_free_ha(oldha);
   ast_free_ha(olddirectmediaha);
   if (!ast_strlen_zero(callback)) { /* build string from peer info */
      char *reg_string;
      if (asprintf(&reg_string, "%s?%s:%s@%s/%s", peer->name, peer->username, !ast_strlen_zero(peer->remotesecret) ? peer->remotesecret : peer->secret, peer->tohost, callback) < 0) {
         ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
      } else if (reg_string) {
         sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
         ast_free(reg_string);
      }
   }
   return peer;
}

static int build_reply_digest	(	struct sip_pvt *	p,
		int	method,
		char *	digest,
		int	digest_len
	)			[static]

Build reply digest.

Returns:

Returns -1 if we have no auth

Note:

Build digest challenge for authentication of registrations and calls Also used for authentication of BYE

Definition at line 18017 of file chan_sip.c.

References append_history, ast_copy_string(), ast_debug, ast_md5_hash(), ast_random(), ast_sockaddr_stringify_host(), ast_strlen_zero(), find_realm_authentication(), secret, sip_methods, and text.

Referenced by reply_digest(), transmit_register(), and transmit_request_with_auth().

{
   char a1[256];
   char a2[256];
   char a1_hash[256];
   char a2_hash[256];
   char resp[256];
   char resp_hash[256];
   char uri[256];
   char opaque[256] = "";
   char cnonce[80];
   const char *username;
   const char *secret;
   const char *md5secret;
   struct sip_auth *auth = NULL; /* Realm authentication */

   if (!ast_strlen_zero(p->domain))
      ast_copy_string(uri, p->domain, sizeof(uri));
   else if (!ast_strlen_zero(p->uri))
      ast_copy_string(uri, p->uri, sizeof(uri));
   else
      snprintf(uri, sizeof(uri), "sip:%s@%s", p->username, ast_sockaddr_stringify_host(&p->sa));

   snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());

   /* Check if we have separate auth credentials */
   if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */
      auth = find_realm_authentication(authl, p->realm); /* If not, global list */

   if (auth) {
      ast_debug(3, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
      username = auth->username;
      secret = auth->secret;
      md5secret = auth->md5secret;
      if (sipdebug)
         ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
   } else {
      /* No authentication, use peer or register= config */
      username = p->authname;
      secret =  p->peersecret;
      md5secret = p->peermd5secret;
   }
   if (ast_strlen_zero(username))   /* We have no authentication */
      return -1;

   /* Calculate SIP digest response */
   snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
   snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
   if (!ast_strlen_zero(md5secret))
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   else
      ast_md5_hash(a1_hash, a1);
   ast_md5_hash(a2_hash, a2);

   p->noncecount++;
   if (!ast_strlen_zero(p->qop))
      snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash);
   else
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
   ast_md5_hash(resp_hash, resp);

   /* only include the opaque string if it's set */
   if (!ast_strlen_zero(p->opaque)) {
     snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
   }

   /* XXX We hard code our qop to "auth" for now.  XXX */
   if (!ast_strlen_zero(p->qop))
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount);
   else
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);

   append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount);

   return 0;
}

static void build_route	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	backwards
	)			[static]

Build route list from Record-Route header.

Definition at line 13123 of file chan_sip.c.

References __get_header(), ast_copy_string(), ast_debug, ast_malloc, ast_strlen_zero(), free_old_route(), get_header(), len(), list_route(), and sip_debug_test_pvt().

Referenced by build_profile(), build_user_routes(), handle_request_invite(), handle_request_subscribe(), and handle_response_invite().

{
   struct sip_route *thishop, *head, *tail;
   int start = 0;
   int len;
   const char *rr, *contact, *c;

   /* Once a persistent route is set, don't fool with it */
   if (p->route && p->route_persistent) {
      ast_debug(1, "build_route: Retaining previous route: <%s>\n", p->route->hop);
      return;
   }

   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }

   /* We only want to create the route set the first time this is called */
   p->route_persistent = 1;
   
   /* Build a tailq, then assign it to p->route when done.
    * If backwards, we add entries from the head so they end up
    * in reverse order. However, we do need to maintain a correct
    * tail pointer because the contact is always at the end.
    */
   head = NULL;
   tail = head;
   /* 1st we pass through all the hops in any Record-Route headers */
   for (;;) {
      /* Each Record-Route header */
      rr = __get_header(req, "Record-Route", &start);
      if (*rr == '\0') {
         break;
      }
      for (; (rr = strchr(rr, '<')) ; rr += len) { /* Each route entry */
         ++rr;
         len = strcspn(rr, ">") + 1;
         /* Make a struct route */
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            /* ast_calloc is not needed because all fields are initialized in this block */
            ast_copy_string(thishop->hop, rr, len);
            ast_debug(2, "build_route: Record-Route hop: <%s>\n", thishop->hop);
            /* Link in */
            if (backwards) {
               /* Link in at head so they end up in reverse order */
               thishop->next = head;
               head = thishop;
               /* If this was the first then it'll be the tail */
               if (!tail) {
                  tail = thishop;
               }
            } else {
               thishop->next = NULL;
               /* Link in at the end */
               if (tail) {
                  tail->next = thishop;
               } else {
                  head = thishop;
               }
               tail = thishop;
            }
         }
      }
   }

   /* Only append the contact if we are dealing with a strict router */
   if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop, ";lr") == NULL) ) {
      /* 2nd append the Contact: if there is one */
      /* Can be multiple Contact headers, comma separated values - we just take the first */
      contact = get_header(req, "Contact");
      if (!ast_strlen_zero(contact)) {
         ast_debug(2, "build_route: Contact hop: %s\n", contact);
         /* Look for <: delimited address */
         c = strchr(contact, '<');
         if (c) {
            /* Take to > */
            ++c;
            len = strcspn(c, ">") + 1;
         } else {
            /* No <> - just take the lot */
            c = contact;
            len = strlen(contact) + 1;
         }
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            /* ast_calloc is not needed because all fields are initialized in this block */
            ast_copy_string(thishop->hop, c, len);
            thishop->next = NULL;
            /* Goes at the end */
            if (tail) {
               tail->next = thishop;
            } else {
               head = thishop;
            }
         }
      }
   }

   /* Store as new route */
   p->route = head;

   /* For debugging dump what we ended up with */
   if (sip_debug_test_pvt(p)) {
      list_route(p->route);
   }
}

static void build_via

(

struct sip_pvt *

p

)

[static]

Build a Via header for a request.

Definition at line 3119 of file chan_sip.c.

References ast_sockaddr_stringify(), ast_test_flag, and get_transport_pvt().

Referenced by __sip_subscribe_mwi_do(), reqprep(), sip_alloc(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_invite(), transmit_register(), and transmit_response_using_temp().

{
   /* Work around buggy UNIDEN UIP200 firmware */
   const char *rport = (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)) ? ";rport" : "";

   /* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
   snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s;branch=z9hG4bK%08x%s",
       get_transport_pvt(p),
       ast_sockaddr_stringify(&p->ourip),
       (int) p->branch, rport);
}

static int cb_extensionstate	(	char *	context,
		char *	exten,
		int	state,
		void *	data
	)			[static]

Callback for the devicestate notification (SUBSCRIBE) support subsystem.

Note:

If you add an "hint" priority to the extension in the dial plan, you will get notifications on device state changes

Definition at line 13498 of file chan_sip.c.

References append_history, AST_EXTENSION_DEACTIVATED, AST_EXTENSION_REMOVED, ast_extension_state2str(), ast_set_flag, ast_test_flag, ast_verb, FALSE, NONE, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), and transmit_state_notify().

Referenced by handle_request_subscribe(), handle_response(), and handle_response_notify().

{
   struct sip_pvt *p = data;

   sip_pvt_lock(p);

   switch(state) {
   case AST_EXTENSION_DEACTIVATED:  /* Retry after a while */
   case AST_EXTENSION_REMOVED:   /* Extension is gone */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);  /* Delete subscription in 32 secs */
      ast_verb(2, "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
      p->stateid = -1;
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "%s", state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated");
      break;
   default: /* Tell user */
      p->laststate = state;
      break;
   }
   if (p->subscribed != NONE) {  /* Only send state NOTIFY if we know the format */
      if (!p->pendinginvite) {
         transmit_state_notify(p, state, 1, FALSE);
      } else {
         /* We already have a NOTIFY sent that is not answered. Queue the state up.
            if many state changes happen meanwhile, we will only send a notification of the last one */
         ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
      }
   }
   ast_verb(2, "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(state), p->username,
         ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : "");

   sip_pvt_unlock(p);

   return 0;
}

static void cc_epa_destructor

(

void *

data

)

[static]

Definition at line 872 of file chan_sip.c.

References ast_free.

{
   struct sip_epa_entry *epa_entry = data;
   struct cc_epa_entry *cc_entry = epa_entry->instance_data;
   ast_free(cc_entry);
}

static void cc_handle_publish_error	(	struct sip_pvt *	pvt,
		const int	resp,
		struct sip_request *	req,
		struct sip_epa_entry *	epa_entry
	)			[static]

Definition at line 18602 of file chan_sip.c.

References ao2_callback, ao2_ref, ast_cc_monitor_failed(), ast_log(), ast_strlen_zero(), FALSE, find_sip_monitor_instance_by_suspension_entry(), get_header(), LOG_WARNING, and transmit_invite().

{
   struct cc_epa_entry *cc_entry = epa_entry->instance_data;
   struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
         find_sip_monitor_instance_by_suspension_entry, epa_entry);
   const char *min_expires;

   if (!monitor_instance) {
      ast_log(LOG_WARNING, "Can't find monitor_instance corresponding to epa_entry %p.\n", epa_entry);
      return;
   }

   if (resp != 423) {
      ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
            "Received error response to our PUBLISH");
      ao2_ref(monitor_instance, -1);
      return;
   }

   /* Allrighty, the other end doesn't like our Expires value. They think it's
    * too small, so let's see if they've provided a more sensible value. If they
    * haven't, then we'll just double our Expires value and see if they like that
    * instead.
    *
    * XXX Ideally this logic could be placed into its own function so that SUBSCRIBE,
    * PUBLISH, and REGISTER could all benefit from the same shared code.
    */
   min_expires = get_header(req, "Min-Expires");
   if (ast_strlen_zero(min_expires)) {
      pvt->expiry *= 2;
      if (pvt->expiry < 0) {
         /* You dork! You overflowed! */
         ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
               "PUBLISH expiry overflowed");
         ao2_ref(monitor_instance, -1);
         return;
      }
   } else if (sscanf(min_expires, "%d", &pvt->expiry) != 1) {
      ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
            "Min-Expires has non-numeric value");
      ao2_ref(monitor_instance, -1);
      return;
   }
   /* At this point, we have most certainly changed pvt->expiry, so try transmitting
    * again
    */
   transmit_invite(pvt, SIP_PUBLISH, FALSE, 0, NULL);
   ao2_ref(monitor_instance, -1);
}

static void change_redirecting_information	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_party_redirecting *	redirecting,
		struct ast_set_party_redirecting *	update_redirecting,
		int	set_call_forward
	)			[static]

update redirecting information for a channel based on headers

Definition at line 18340 of file chan_sip.c.

References ast_debug, ast_free, AST_REDIRECTING_REASON_UNCONDITIONAL, ast_strdup, ast_strlen_zero(), ast_party_redirecting::from, ast_set_party_redirecting::from, get_header(), get_rdnis(), ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_moved_contact(), ast_party_redirecting::reason, ast_party_name::str, ast_party_number::str, ast_party_id::tag, ast_set_party_redirecting::to, ast_party_redirecting::to, ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_request_invite(), handle_response(), and handle_response_invite().

{
   char *redirecting_from_name = NULL;
   char *redirecting_from_number = NULL;
   char *redirecting_to_name = NULL;
   char *redirecting_to_number = NULL;
   int reason = AST_REDIRECTING_REASON_UNCONDITIONAL;
   int is_response = req->method == SIP_RESPONSE;
   int res = 0;

   res = get_rdnis(p, req, &redirecting_from_name, &redirecting_from_number, &reason);
   if (res == -1) {
      if (is_response) {
         get_name_and_number(get_header(req, "TO"), &redirecting_from_name, &redirecting_from_number);
      } else {
         return;
      }
   }

   /* At this point, all redirecting "from" info should be filled in appropriately
    * on to the "to" info
    */

   if (is_response) {
      parse_moved_contact(p, req, &redirecting_to_name, &redirecting_to_number, set_call_forward);
   } else {
      get_name_and_number(get_header(req, "TO"), &redirecting_to_name, &redirecting_to_number);
   }

   if (!ast_strlen_zero(redirecting_from_number)) {
      ast_debug(3, "Got redirecting from number %s\n", redirecting_from_number);
      update_redirecting->from.number = 1;
      redirecting->from.number.valid = 1;
      ast_free(redirecting->from.number.str);
      redirecting->from.number.str = redirecting_from_number;
   }
   if (!ast_strlen_zero(redirecting_from_name)) {
      ast_debug(3, "Got redirecting from name %s\n", redirecting_from_name);
      update_redirecting->from.name = 1;
      redirecting->from.name.valid = 1;
      ast_free(redirecting->from.name.str);
      redirecting->from.name.str = redirecting_from_name;
   }
   if (!ast_strlen_zero(p->cid_tag)) {
      ast_free(redirecting->from.tag);
      redirecting->from.tag = ast_strdup(p->cid_tag);
      ast_free(redirecting->to.tag);
      redirecting->to.tag = ast_strdup(p->cid_tag);
   }
   if (!ast_strlen_zero(redirecting_to_number)) {
      ast_debug(3, "Got redirecting to number %s\n", redirecting_to_number);
      update_redirecting->to.number = 1;
      redirecting->to.number.valid = 1;
      ast_free(redirecting->to.number.str);
      redirecting->to.number.str = redirecting_to_number;
   }
   if (!ast_strlen_zero(redirecting_to_name)) {
      ast_debug(3, "Got redirecting to name %s\n", redirecting_from_number);
      update_redirecting->to.name = 1;
      redirecting->to.name.valid = 1;
      ast_free(redirecting->to.name.str);
      redirecting->to.name.str = redirecting_to_name;
   }
   redirecting->reason = reason;
}

static void change_t38_state	(	struct sip_pvt *	p,
		int	state
	)			[static]

Change the T38 state on a SIP dialog.

Definition at line 4669 of file chan_sip.c.

References AST_CONTROL_T38_PARAMETERS, ast_debug, ast_queue_control_data(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_TERMINATED, ast_udptl_get_far_max_ifp(), ast_udptl_set_tag(), ast_control_t38_parameters::max_ifp, ast_channel::name, and ast_control_t38_parameters::request_response.

Referenced by handle_response_invite(), interpret_t38_parameters(), process_sdp(), and sip_t38_abort().

{
   int old = p->t38.state;
   struct ast_channel *chan = p->owner;
   struct ast_control_t38_parameters parameters = { .request_response = 0 };

   /* Don't bother changing if we are already in the state wanted */
   if (old == state)
      return;

   p->t38.state = state;
   ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");

   /* If no channel was provided we can't send off a control frame */
   if (!chan)
      return;

   /* Given the state requested and old state determine what control frame we want to queue up */
   switch (state) {
   case T38_PEER_REINVITE:
      parameters = p->t38.their_parms;
      parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
      parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
      ast_udptl_set_tag(p->udptl, "SIP/%s", p->username);
      break;
   case T38_ENABLED:
      parameters = p->t38.their_parms;
      parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
      parameters.request_response = AST_T38_NEGOTIATED;
      ast_udptl_set_tag(p->udptl, "SIP/%s", p->username);
      break;
   case T38_DISABLED:
      if (old == T38_ENABLED) {
         parameters.request_response = AST_T38_TERMINATED;
      } else if (old == T38_LOCAL_REINVITE) {
         parameters.request_response = AST_T38_REFUSED;
      }
      break;
   case T38_LOCAL_REINVITE:
      /* wait until we get a peer response before responding to local reinvite */
      break;
   }

   /* Woot we got a message, create a control frame and send it on! */
   if (parameters.request_response)
      ast_queue_control_data(chan, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
}

static enum check_auth_result check_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const char *	username,
		const char *	secret,
		const char *	md5secret,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable,
		int	ignore
	)			[static]

Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).

Returns:

0 on success, non-zero on error

XXX

Todo:

need a better return code here

XXX

Todo:

need a better return code here

Definition at line 13252 of file chan_sip.c.

References append_history, ast_copy_string(), AST_DYNSTR_BUILD_FAILED, ast_log(), ast_md5_hash(), ast_skip_blanks(), ast_str_set(), ast_str_thread_get(), ast_strlen_zero(), auth_headers(), check_auth_buf, CHECK_AUTH_BUF_INITLEN, FALSE, get_header(), LOG_NOTICE, LOG_WARNING, S_OR, set_nonce_randdata(), sip_methods, sip_scheddestroy(), strsep(), text, transmit_response_with_auth(), and TRUE.

Referenced by check_peer_ok(), and register_verify().

{
   const char *response;
   char *reqheader, *respheader;
   const char *authtoken;
   char a1_hash[256];
   char resp_hash[256]="";
   char *c;
   int  wrongnonce = FALSE;
   int  good_response;
   const char *usednonce = p->randdata;
   struct ast_str *buf;
   int res;

   /* table of recognised keywords, and their value in the digest */
   enum keys { K_RESP, K_URI, K_USER, K_NONCE, K_LAST };
   struct x {
      const char *key;
      const char *s;
   } *i, keys[] = {
      [K_RESP] = { "response=", "" },
      [K_URI] = { "uri=", "" },
      [K_USER] = { "username=", "" },
      [K_NONCE] = { "nonce=", "" },
      [K_LAST] = { NULL, NULL}
   };

   /* Always OK if no secret */
   if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret))
      return AUTH_SUCCESSFUL;

   /* Always auth with WWW-auth since we're NOT a proxy */
   /* Using proxy-auth in a B2BUA may block proxy authorization in the same transaction */
   response = "401 Unauthorized";

   /*
    * Note the apparent swap of arguments below, compared to other
    * usages of auth_headers().
    */
   auth_headers(WWW_AUTH, &respheader, &reqheader);

   authtoken =  get_header(req, reqheader);  
   if (ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) {
      /* This is a retransmitted invite/register/etc, don't reconstruct authentication
         information */
      if (!reliable) {
         /* Resend message if this was NOT a reliable delivery.   Otherwise the
            retransmission should get it */
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
         /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
      return AUTH_CHALLENGE_SENT;
   } else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
      /* We have no auth, so issue challenge and request authentication */
      set_nonce_randdata(p, 1); /* Create nonce for challenge */
      transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   }

   /* --- We have auth, so check it */

   /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting
      an example in the spec of just what it is you're doing a hash on. */

   if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
   }

   /* Make a copy of the response and parse it */
   res = ast_str_set(&buf, 0, "%s", authtoken);

   if (res == AST_DYNSTR_BUILD_FAILED) {
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
   }

   c = buf->str;

   while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         const char *separator = ",";  /* default */

         if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
            continue;
         }
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') { /* in quotes. Skip first and look for last */
            c++;
            separator = "\"";
         }
         i->s = c;
         strsep(&c, separator);
         break;
      }
      if (i->key == NULL) { /* not found, jump after space or comma */
         strsep(&c, " ,");
      }
   }

   /* Verify that digest username matches  the username we auth as */
   if (strcmp(username, keys[K_USER].s)) {
      ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n",
         username, keys[K_USER].s);
      /* Oops, we're trying something here */
      return AUTH_USERNAME_MISMATCH;
   }

   /* Verify nonce from request matches our nonce, and the nonce has not already been responded to.
    * If this check fails, send 401 with new nonce */
   if (strcasecmp(p->randdata, keys[K_NONCE].s) || p->stalenonce) { /* XXX it was 'n'casecmp ? */
      wrongnonce = TRUE;
      usednonce = keys[K_NONCE].s;
   } else {
      p->stalenonce = 1; /* now, since the nonce has a response, mark it as stale so it can't be sent or responded to again */
   }

   if (!ast_strlen_zero(md5secret)) {
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   } else {
      char a1[256];

      snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
      ast_md5_hash(a1_hash, a1);
   }

   /* compute the expected response to compare with what we received */
   {
      char a2[256];
      char a2_hash[256];
      char resp[256];

      snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text,
            S_OR(keys[K_URI].s, uri));
      ast_md5_hash(a2_hash, a2);
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash);
      ast_md5_hash(resp_hash, resp);
   }

   good_response = keys[K_RESP].s &&
         !strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash));
   if (wrongnonce) {
      if (good_response) {
         if (sipdebug)
            ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", get_header(req, "From"));
         /* We got working auth token, based on stale nonce . */
         set_nonce_randdata(p, 0);
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, TRUE);
      } else {
         /* Everything was wrong, so give the device one more try with a new challenge */
         if (!req->ignore) {
            if (sipdebug) {
               ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", get_header(req, "To"));
            }
            set_nonce_randdata(p, 1);
         } else {
            if (sipdebug) {
               ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", get_header(req, "To"));
            }
         }
         transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE);
      }

      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   }
   if (good_response) {
      append_history(p, "AuthOK", "Auth challenge successful for %s", username);
      return AUTH_SUCCESSFUL;
   }

   /* Ok, we have a bad username/secret pair */
   /* Tell the UAS not to re-send this authentication data, because
      it will continue to fail
   */

   return AUTH_SECRET_FAILED;
}

static enum check_auth_result check_peer_ok	(	struct sip_pvt *	p,
		char *	of,
		struct sip_request *	req,
		int	sipmethod,
		struct ast_sockaddr *	addr,
		struct sip_peer **	authpeer,
		enum xmittype	reliable,
		char *	calleridname,
		char *	uri2
	)			[static]

Validate device authentication.

Definition at line 14800 of file chan_sip.c.

References accountcode, ao2_t_ref, ast_apply_ha(), ast_cc_copy_config_params(), ast_copy_flags, ast_debug, ast_is_shrinkable_phonenumber(), ast_rtp_codecs_packetization_set(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_set_flag, ast_shrink_phone_number(), ast_sockaddr_stringify(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose, check_auth(), cid_name, cid_num, context, copy_vars(), debug, dialog_initialize_rtp(), do_setnat(), FALSE, find_peer(), get_rpid(), language, mohinterpret, mohsuggest, parkinglot, set_pvt_allowed_methods(), set_t38_capabilities(), sip_debug_test_addr(), TRUE, and unref_peer().

Referenced by check_user_full().

{
   enum check_auth_result res;
   int debug = sip_debug_test_addr(addr);
   struct sip_peer *peer;

   if (sipmethod == SIP_SUBSCRIBE) {
      /* For subscribes, match on device name only; for other methods,
      * match on IP address-port of the incoming request.
      */
      peer = find_peer(of, NULL, TRUE, FINDALLDEVICES, FALSE, 0);
   } else {
      /* First find devices based on username (avoid all type=peer's) */
      peer = find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0);

      /* Then find devices based on IP */
      if (!peer) {
         peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);
      }
   }

   if (!peer) {
      if (debug) {
         ast_verbose("No matching peer for '%s' from '%s'\n",
            of, ast_sockaddr_stringify(&p->recv));
      }
      return AUTH_DONT_KNOW;
   }

   if (!ast_apply_ha(peer->ha, addr)) {
      ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
      unref_peer(peer, "unref_peer: check_peer_ok: from find_peer call, early return of AUTH_ACL_FAILED");
      return AUTH_ACL_FAILED;
   }
   if (debug)
      ast_verbose("Found peer '%s' for '%s' from %s\n",
         peer->name, of, ast_sockaddr_stringify(&p->recv));

   /* XXX what about p->prefs = peer->prefs; ? */
   /* Set Frame packetization */
   if (p->rtp) {
      ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
      p->autoframing = peer->autoframing;
   }

   /* Take the peer */
   ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->udptl) {
      p->t38_maxdatagram = peer->t38_maxdatagram;
      set_t38_capabilities(p);
   }

   /* Copy SIP extensions profile to peer */
   /* XXX is this correct before a successful auth ? */
   if (p->sipoptions)
      peer->sipoptions = p->sipoptions;

   do_setnat(p);

   ast_string_field_set(p, peersecret, peer->secret);
   ast_string_field_set(p, peermd5secret, peer->md5secret);
   ast_string_field_set(p, subscribecontext, peer->subscribecontext);
   ast_string_field_set(p, mohinterpret, peer->mohinterpret);
   ast_string_field_set(p, mohsuggest, peer->mohsuggest);
   ast_string_field_set(p, parkinglot, peer->parkinglot);
   ast_string_field_set(p, engine, peer->engine);
   p->disallowed_methods = peer->disallowed_methods;
   set_pvt_allowed_methods(p, req);
   ast_cc_copy_config_params(p->cc_params, peer->cc_params);
   if (peer->callingpres)  /* Peer calling pres setting will override RPID */
      p->callingpres = peer->callingpres;
   if (peer->maxms && peer->lastms)
      p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      p->timer_t1 = peer->timer_t1;

   /* Set timer B to control transaction timeouts */
   if (peer->timer_b)
      p->timer_b = peer->timer_b;
   else
      p->timer_b = 64 * p->timer_t1;

   if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
      /* Pretend there is no required authentication */
      ast_string_field_set(p, peersecret, NULL);
      ast_string_field_set(p, peermd5secret, NULL);
   }
   if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, req->ignore))) {
      ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
      ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
      ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
      /* If we have a call limit, set flag */
      if (peer->call_limit)
         ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
      ast_string_field_set(p, peername, peer->name);
      ast_string_field_set(p, authname, peer->name);

      if (sipmethod == SIP_INVITE) {
         /* copy channel vars */
         p->chanvars = copy_vars(peer->chanvars);
      }

      if (authpeer) {
         ao2_t_ref(peer, 1, "copy pointer into (*authpeer)");
         (*authpeer) = peer;  /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
      }

      if (!ast_strlen_zero(peer->username)) {
         ast_string_field_set(p, username, peer->username);
         /* Use the default username for authentication on outbound calls */
         /* XXX this takes the name from the caller... can we override ? */
         ast_string_field_set(p, authname, peer->username);
      }
      if (!get_rpid(p, req)) {
         if (!ast_strlen_zero(peer->cid_num)) {
            char *tmp = ast_strdupa(peer->cid_num);
            if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp))
               ast_shrink_phone_number(tmp);
            ast_string_field_set(p, cid_num, tmp);
         }
         if (!ast_strlen_zero(peer->cid_name))
            ast_string_field_set(p, cid_name, peer->cid_name);
         if (!ast_strlen_zero(peer->cid_tag))
            ast_string_field_set(p, cid_tag, peer->cid_tag);
         if (peer->callingpres)
            p->callingpres = peer->callingpres;
      }
      ast_string_field_set(p, fullcontact, peer->fullcontact);
      if (!ast_strlen_zero(peer->context))
         ast_string_field_set(p, context, peer->context);
      ast_string_field_set(p, peersecret, peer->secret);
      ast_string_field_set(p, peermd5secret, peer->md5secret);
      ast_string_field_set(p, language, peer->language);
      ast_string_field_set(p, accountcode, peer->accountcode);
      p->amaflags = peer->amaflags;
      p->callgroup = peer->callgroup;
      p->pickupgroup = peer->pickupgroup;
      p->capability = peer->capability;
      p->prefs = peer->prefs;
      p->jointcapability = peer->capability;
      if (peer->maxforwards > 0) {
         p->maxforwards = peer->maxforwards;
      }
      if (p->peercapability)
         p->jointcapability &= p->peercapability;
      p->maxcallbitrate = peer->maxcallbitrate;
      if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
          (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
         p->noncodeccapability |= AST_RTP_DTMF;
      else
         p->noncodeccapability &= ~AST_RTP_DTMF;
      p->jointnoncodeccapability = p->noncodeccapability;
      if (!dialog_initialize_rtp(p)) {
         if (p->rtp) {
            ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
            p->autoframing = peer->autoframing;
         }
      } else {
         res = AUTH_RTP_FAILED;
      }
   }
   unref_peer(peer, "check_peer_ok: unref_peer: tossing temp ptr to peer from find_peer");
   return res;
}

static void check_pendings

(

struct sip_pvt *

p

)

[static]

Check pending actions on SIP call.

Note:

both sip_pvt and sip_pvt's owner channel (if present) must be locked for this function.

Definition at line 18526 of file chan_sip.c.

References ast_clear_flag, ast_debug, AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_test_flag, FALSE, sip_scheddestroy(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), and TRUE.

Referenced by handle_incoming(), handle_response_invite(), and sip_reinvite_retry().

{
   if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
      /* if we can't BYE, then this is really a pending CANCEL */
      if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
         transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
         /* Actually don't destroy us yet, wait for the 487 on our original
            INVITE, but do set an autodestruct just in case we never get it. */
      else {
         /* We have a pending outbound invite, don't send something
            new in-transaction */
         if (p->pendinginvite)
            return;

         if (p->owner) {
            ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
         }
         /* Perhaps there is an SD change INVITE outstanding */
         transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
      }
      ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);   
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
      /* if we can't REINVITE, hold it for later */
      if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) {
         ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
      } else {
         ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
         /* Didn't get to reinvite yet, so do it now */
         transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
         ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); 
      }
   }
}

static void check_rtp_timeout	(	struct sip_pvt *	dialog,
		time_t	t
	)			[static]

helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked

Todo:

Check video RTP keepalives

Do we need to move the lastrtptx to the RTP structure to have one for audio and one for video? It really does belong to the RTP structure.

Definition at line 24169 of file chan_sip.c.

References ast_channel_trylock, ast_channel_unlock, ast_log(), ast_rtp_instance_get_hold_timeout(), ast_rtp_instance_get_timeout(), ast_rtp_instance_set_hold_timeout(), ast_rtp_instance_set_timeout(), ast_sockaddr_isnull(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, ast_test_flag, LOG_NOTICE, sip_pvt_lock, and sip_pvt_unlock.

Referenced by dialog_needdestroy().

{
   /* If we have no RTP or no active owner, no need to check timers */
   if (!dialog->rtp || !dialog->owner)
      return;
   /* If the call is not in UP state or redirected outside Asterisk, no need to check timers */

   if (dialog->owner->_state != AST_STATE_UP || !ast_sockaddr_isnull(&dialog->redirip))
      return;

   /* If the call is involved in a T38 fax session do not check RTP timeout */
   if (dialog->t38.state == T38_ENABLED)
      return;

   /* If we have no timers set, return now */
   if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
      return;
   }

   /*! \todo Check video RTP keepalives

      Do we need to move the lastrtptx to the RTP structure to have one for audio and one
      for video? It really does belong to the RTP structure.
   */

   /* Check AUDIO RTP timers */
   if (dialog->lastrtprx && (ast_rtp_instance_get_timeout(dialog->rtp) || ast_rtp_instance_get_hold_timeout(dialog->rtp)) && (t > dialog->lastrtprx + ast_rtp_instance_get_timeout(dialog->rtp))) {
      if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
         /* Needs a hangup */
         if (ast_rtp_instance_get_timeout(dialog->rtp)) {
            while (dialog->owner && ast_channel_trylock(dialog->owner)) {
               sip_pvt_unlock(dialog);
               usleep(1);
               sip_pvt_lock(dialog);
            }
            if (!dialog->owner) {
               return; /* channel hangup can occur during deadlock avoidance. */
            }
            ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
               dialog->owner->name, (long) (t - dialog->lastrtprx));
            /* Issue a softhangup */
            ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
            ast_channel_unlock(dialog->owner);
            /* forget the timeouts for this call, since a hangup
               has already been requested and we don't want to
               repeatedly request hangups
            */
            ast_rtp_instance_set_timeout(dialog->rtp, 0);
            ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
            if (dialog->vrtp) {
               ast_rtp_instance_set_timeout(dialog->vrtp, 0);
               ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
            }
         }
      }
   }
}

static int check_sip_domain	(	const char *	domain,
		char *	context,
		size_t	len
	)			[static]

check_sip_domain: Check if domain part of uri is local to our server

Definition at line 25315 of file chan_sip.c.

References ast_copy_string(), AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, and ast_strlen_zero().

Referenced by func_check_sipdomain(), get_destination(), get_realm(), handle_request_refer(), and register_verify().

{
   struct domain *d;
   int result = 0;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_TRAVERSE(&domain_list, d, list) {
      if (strcasecmp(d->domain, domain))
         continue;

      if (len && !ast_strlen_zero(d->context))
         ast_copy_string(context, d->context, len);
      
      result = 1;
      break;
   }
   AST_LIST_UNLOCK(&domain_list);

   return result;
}

static int check_user	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable,
		struct ast_sockaddr *	addr
	)			[static]

Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.

Definition at line 15100 of file chan_sip.c.

References check_user_full().

Referenced by handle_request_options(), and handle_request_publish().

{
   return check_user_full(p, req, sipmethod, uri, reliable, addr, NULL);
}

static enum check_auth_result check_user_full	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable,
		struct ast_sockaddr *	addr,
		struct sip_peer **	authpeer
	)			[static]

Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.

Returns:

0 on success, non-zero on failure

Definition at line 14977 of file chan_sip.c.

References ast_copy_string(), ast_is_shrinkable_phonenumber(), ast_log(), ast_set_flag, ast_shrink_phone_number(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_contact(), check_peer_ok(), cid_name, cid_num, dialog_initialize_rtp(), dummy(), exten, get_header(), get_rpid(), LOG_ERROR, LOG_NOTICE, sip_cfg, SIP_PEDANTIC_DECODE, strsep(), and terminate_uri().

Referenced by check_user(), handle_request_invite(), and handle_request_subscribe().

{
   char from[256] = { 0, };
   char *dummy = NULL;  /* dummy return value for parse_uri */
   char *domain = NULL; /* dummy return value for parse_uri */
   char *of;
   enum check_auth_result res = AUTH_DONT_KNOW;
   char calleridname[50];
   char *uri2 = ast_strdupa(uri);

   terminate_uri(uri2); /* trim extra stuff */

   ast_copy_string(from, get_header(req, "From"), sizeof(from));
   /* XXX here tries to map the username for invite things */
   memset(calleridname, 0, sizeof(calleridname));

   /* strip the display-name portion off the beginning of the FROM header. */
   if (!(of = (char *) get_calleridname(from, calleridname, sizeof(calleridname)))) {
      ast_log(LOG_ERROR, "FROM header can not be parsed \n");
      return res;
   }

   if (calleridname[0])
      ast_string_field_set(p, cid_name, calleridname);

   if (ast_strlen_zero(p->exten)) {
      char *t = uri2;
      if (!strncasecmp(t, "sip:", 4))
         t+= 4;
      else if (!strncasecmp(t, "sips:", 5))
         t += 5;
      ast_string_field_set(p, exten, t);
      t = strchr(p->exten, '@');
      if (t)
         *t = '\0';

      if (ast_strlen_zero(p->our_contact))
         build_contact(p);
   }

   of = get_in_brackets(of);

   /* save the URI part of the From header */
   ast_string_field_set(p, from, of);

   /* ignore all fields but name */
   if (parse_uri(of, "sip:,sips:", &of, &dummy, &domain, NULL)) {
      ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
   }

   SIP_PEDANTIC_DECODE(of);
   SIP_PEDANTIC_DECODE(domain);

   if (ast_strlen_zero(of)) {
      /* XXX note: the original code considered a missing @host
       * as a username-only URI. The SIP RFC (19.1.1) says that
       * this is wrong, and it should be considered as a domain-only URI.
       * For backward compatibility, we keep this block, but it is
       * really a mistake and should go away.
       */
      of = domain;
   } else {
      char *tmp = ast_strdupa(of);
      /* We need to be able to handle auth-headers looking like
         <sip:8164444422;phone-context=+1@1.2.3.4:5060;user=phone;tag=SDadkoa01-gK0c3bdb43>
      */
      tmp = strsep(&tmp, ";");
      if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp))
         ast_shrink_phone_number(tmp);
      ast_string_field_set(p, cid_num, tmp);
   }

   if (global_match_auth_username) {
      /*
       * XXX This is experimental code to grab the search key from the
       * Auth header's username instead of the 'From' name, if available.
       * Do not enable this block unless you understand the side effects (if any!)
       * Note, the search for "username" should be done in a more robust way.
       * Note2, at the moment we check both fields, though maybe we should
       * pick one or another depending on the request ? XXX
       */
      const char *hdr = get_header(req, "Authorization");
      if (ast_strlen_zero(hdr))
         hdr = get_header(req, "Proxy-Authorization");

      if ( !ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\"")) ) {
         ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
         of = from;
         of = strsep(&of, "\"");
      }
   }

   res = check_peer_ok(p, of, req, sipmethod, addr,
         authpeer, reliable, calleridname, uri2);
   if (res != AUTH_DONT_KNOW)
      return res;

   /* Finally, apply the guest policy */
   if (sip_cfg.allowguest) {
      get_rpid(p, req);
      if (!dialog_initialize_rtp(p)) {
         res = AUTH_SUCCESSFUL;
      } else {
         res = AUTH_RTP_FAILED;
      }
   } else if (sip_cfg.alwaysauthreject)
      res = AUTH_FAKE_AUTH; /* reject with fake authorization request */
   else
      res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */


   if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT)) {
      ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
   }

   return res;
}

static void check_via	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

check Via: header for hostname, port and rport request/answer

Definition at line 14739 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_set_flag, ast_skip_blanks(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_verbose, get_header(), LOG_WARNING, sip_debug_test_pvt(), sip_nat_mode(), and sip_real_dst().

Referenced by handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_register(), and handle_request_subscribe().

{
   char via[512];
   char *c, *maddr;
   struct ast_sockaddr tmp;
   uint16_t port;

   ast_copy_string(via, get_header(req, "Via"), sizeof(via));

   /* Work on the leftmost value of the topmost Via header */
   c = strchr(via, ',');
   if (c)
      *c = '\0';

   /* Check for rport */
   c = strstr(via, ";rport");
   if (c && (c[6] != '=')) { /* rport query, not answer */
      ast_set_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT);
      ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
   }

   /* Check for maddr */
   maddr = strstr(via, "maddr=");
   if (maddr) {
      maddr += 6;
      c = maddr + strspn(maddr, "abcdefghijklmnopqrstuvwxyz"
                      "ABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789-.:[]");
      *c = '\0';
   }

   c = strchr(via, ';');
   if (c)
      *c = '\0';

   c = strchr(via, ' ');
   if (c) {
      *c = '\0';
      c = ast_skip_blanks(c+1);
      if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
         ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
         return;
      }

      if (maddr && ast_sockaddr_resolve_first(&p->sa, maddr, 0)) {
         p->sa = p->recv;
      }

      ast_sockaddr_resolve_first(&tmp, c, 0);
      port = ast_sockaddr_port(&tmp);
      ast_sockaddr_set_port(&p->sa,
                  port != 0 ? port : STANDARD_SIP_PORT);

      if (sip_debug_test_pvt(p)) {
         ast_verbose("Sending to %s (%s)\n",
                ast_sockaddr_stringify(sip_real_dst(p)),
                sip_nat_mode(p));
      }
   }
}

static attribute_unused void check_via_response	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

check received= and rport= in a SIP response. If we get a response with received= and/or rport= in the Via: line, use them as 'p->ourip' (see RFC 3581 for rport, and RFC 3261 for received). Using these two fields SIP can produce the correct address and port in the SIP headers without the need for STUN. The address part is also reused for the media sessions. Note that ast_sip_ouraddrfor() still rewrites p->ourip if you specify externaddr/seternaddr/.

Definition at line 14709 of file chan_sip.c.

References ast_copy_string(), ast_parse_arg(), ast_sockaddr_set_port, get_header(), PARSE_ADDR, and strsep().

{
   char via[256];
   char *cur, *opts;

   ast_copy_string(via, get_header(req, "Via"), sizeof(via));

   /* Work on the leftmost value of the topmost Via header */
   opts = strchr(via, ',');
   if (opts)
      *opts = '\0';

   /* parse all relevant options */
   opts = strchr(via, ';');
   if (!opts)
      return;  /* no options to parse */
   *opts++ = '\0';
   while ( (cur = strsep(&opts, ";")) ) {
      if (!strncmp(cur, "rport=", 6)) {
         int port = strtol(cur+6, NULL, 10);
         /* XXX add error checking */
         ast_sockaddr_set_port(&p->ourip, port);
      } else if (!strncmp(cur, "received=", 9)) {
         if (ast_parse_arg(cur + 9, PARSE_ADDR, &p->ourip))
            ;  /* XXX add error checking */
      }
   }
}

static void cleanup_stale_contexts	(	char *	new,
		char *	old
	)			[static]

Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.

Definition at line 15783 of file chan_sip.c.

References ast_context_destroy(), ast_context_find(), ast_copy_string(), AST_MAX_CONTEXT, and strsep().

Referenced by config_parse_variables(), and reload_config().

{
   char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT];

   while ((oldcontext = strsep(&old, "&"))) {
      stalecontext = '\0';
      ast_copy_string(newlist, new, sizeof(newlist));
      stringp = newlist;
      while ((newcontext = strsep(&stringp, "&"))) {
         if (!strcmp(newcontext, oldcontext)) {
            /* This is not the context you're looking for */
            stalecontext = '\0';
            break;
         } else if (strcmp(newcontext, oldcontext)) {
            stalecontext = oldcontext;
         }
         
      }
      if (stalecontext)
         ast_context_destroy(ast_context_find(stalecontext), "SIP");
   }
}

static void clear_peer_mailboxes

(

struct sip_peer *

peer

)

[static]

Destroy all peer-related mailbox subscriptions

Definition at line 4267 of file chan_sip.c.

References AST_LIST_REMOVE_HEAD, destroy_mailbox(), and mailbox.

Referenced by set_peer_defaults(), and sip_destroy_peer().

{
   struct sip_mailbox *mailbox;

   while ((mailbox = AST_LIST_REMOVE_HEAD(&peer->mailboxes, entry)))
      destroy_mailbox(mailbox);
}

static int clear_realm_authentication

(

struct sip_auth *

authlist

)

[static]

Clear realm authentication list (at reload).

Definition at line 25404 of file chan_sip.c.

References ast_free.

Referenced by build_peer(), reload_config(), sip_destroy_peer(), and unload_module().

{
   struct sip_auth *a = authlist;
   struct sip_auth *b;

   while (a) {
      b = a;
      a = a->next;
      ast_free(b);
   }

   return 1;
}

static void clear_sip_domains

(

void

)

[static]

Clear our domain list (at reload).

Definition at line 25337 of file chan_sip.c.

References ast_free, AST_LIST_LOCK, AST_LIST_REMOVE_HEAD, and AST_LIST_UNLOCK.

Referenced by reload_config(), and unload_module().

{
   struct domain *d;

   AST_LIST_LOCK(&domain_list);
   while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list)))
      ast_free(d);
   AST_LIST_UNLOCK(&domain_list);
}

static char * complete_sip_peer	(	const char *	word,
		int	state,
		int	flags2
	)			[static]

Do completion on peer name.

Definition at line 17179 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, and unref_peer().

Referenced by complete_sip_show_peer(), complete_sipnotify(), sip_do_debug(), and sip_prune_realtime().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator i = ao2_iterator_init(peers, 0);
   struct sip_peer *peer;

   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, peer->name, wordlen) &&
            (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
            ++which > state)
         result = ast_strdup(peer->name);
      unref_peer(peer, "toss iterator peer ptr before break");
      if (result) {
         break;
      }
   }
   ao2_iterator_destroy(&i);
   return result;
}

static char * complete_sip_registered_peer	(	const char *	word,
		int	state,
		int	flags2
	)			[static]

Do completion on registered peer name.

Definition at line 17203 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, and unref_peer().

Referenced by complete_sip_unregister().

{
       char *result = NULL;
       int wordlen = strlen(word);
       int which = 0;
       struct ao2_iterator i;
       struct sip_peer *peer;
       
       i = ao2_iterator_init(peers, 0);
       while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
          if (!strncasecmp(word, peer->name, wordlen) &&
         (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
         ++which > state && peer->expire > 0)
             result = ast_strdup(peer->name);
          if (result) {
             unref_peer(peer, "toss iterator peer ptr before break");
             break;
          }
          unref_peer(peer, "toss iterator peer ptr");
       }
       ao2_iterator_destroy(&i);
       return result;
}

static char * complete_sip_show_history	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show history' CLI.

Definition at line 17228 of file chan_sip.c.

References complete_sipch().

Referenced by sip_show_history().

{
   if (pos == 3)
      return complete_sipch(line, word, pos, state);

   return NULL;
}

static char * complete_sip_show_peer	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show peer' CLI.

Definition at line 17237 of file chan_sip.c.

References complete_sip_peer().

Referenced by sip_qualify_peer(), and sip_show_peer().

{
   if (pos == 3) {
      return complete_sip_peer(word, state, 0);
   }

   return NULL;
}

static char* complete_sip_show_user	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show user' CLI.

Definition at line 16480 of file chan_sip.c.

References complete_sip_user().

Referenced by sip_show_user().

{
   if (pos == 3)
      return complete_sip_user(word, state);

   return NULL;
}

static char * complete_sip_unregister	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip unregister' CLI.

Definition at line 17247 of file chan_sip.c.

References complete_sip_registered_peer().

Referenced by sip_unregister().

{
       if (pos == 2)
               return complete_sip_registered_peer(word, state, 0);

       return NULL;
}

static char* complete_sip_user	(	const char *	word,
		int	state
	)			[static]

Do completion on user name.

Definition at line 16453 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_lock, ao2_unlock, ast_strdup, and unref_peer().

Referenced by complete_sip_show_user().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator user_iter;
   struct sip_peer *user;

   user_iter = ao2_iterator_init(peers, 0);
   while ((user = ao2_iterator_next(&user_iter))) {
      ao2_lock(user);
      if (!(user->type & SIP_TYPE_USER)) {
         ao2_unlock(user);
         unref_peer(user, "complete sip user");
         continue;
      }
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, user->name, wordlen) && ++which > state) {
         result = ast_strdup(user->name);
      }
      ao2_unlock(user);
      unref_peer(user, "complete sip user");
   }
   ao2_iterator_destroy(&user_iter);
   return result;
}

static char* complete_sipch	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.

Definition at line 17149 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, sip_pvt_lock, and sip_pvt_unlock.

Referenced by complete_sip_show_history(), and sip_show_channel().

{
   int which=0;
   struct sip_pvt *cur;
   char *c = NULL;
   int wordlen = strlen(word);
   struct ao2_iterator i;

   if (pos != 3) {
      return NULL;
   }

   i = ao2_iterator_init(dialogs, 0);
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);
      if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) {
         c = ast_strdup(cur->callid);
         sip_pvt_unlock(cur);
         dialog_unref(cur, "drop ref in iterator loop break");
         break;
      }
      sip_pvt_unlock(cur);
      dialog_unref(cur, "drop ref in iterator loop");
   }
   ao2_iterator_destroy(&i);
   return c;
}

static char * complete_sipnotify	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)			[static]

Support routine for 'sip notify' CLI.

Definition at line 17256 of file chan_sip.c.

References ast_category_browse(), ast_strdup, and complete_sip_peer().

Referenced by sip_cli_notify().

{
   char *c = NULL;

   if (pos == 2) {
      int which = 0;
      char *cat = NULL;
      int wordlen = strlen(word);

      /* do completion for notify type */

      if (!notify_types)
         return NULL;
      
      while ( (cat = ast_category_browse(notify_types, cat)) ) {
         if (!strncasecmp(word, cat, wordlen) && ++which > state) {
            c = ast_strdup(cat);
            break;
         }
      }
      return c;
   }

   if (pos > 2)
      return complete_sip_peer(word, state, 0);

   return NULL;
}

static int construct_pidf_body	(	enum sip_cc_publish_state	state,
		char *	pidf_body,
		size_t	size,
		const char *	presentity
	)			[static]

Definition at line 1901 of file chan_sip.c.

References ast_copy_string(), ast_str_alloca, ast_str_append(), ast_str_buffer(), and generate_random_string().

Referenced by handle_cc_notify(), sip_cc_monitor_suspend(), and sip_cc_monitor_unsuspend().

{
   struct ast_str *body = ast_str_alloca(size);
   char tuple_id[32];

   generate_random_string(tuple_id, sizeof(tuple_id));

   /* We'll make this a bare-bones pidf body. In state_notify_build_xml, the PIDF
    * body gets a lot more extra junk that isn't necessary, so we'll leave it out here.
    */
   ast_str_append(&body, 0, "<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n");
   /* XXX The entity attribute is currently set to the peer name associated with the
    * dialog. This is because we currently only call this function for call-completion
    * PUBLISH bodies. In such cases, the entity is completely disregarded. For other
    * event packages, it may be crucial to have a proper URI as the presentity so this
    * should be revisited as support is expanded.
    */
   ast_str_append(&body, 0, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" entity=\"%s\">\n", presentity);
   ast_str_append(&body, 0, "<tuple id=\"%s\">\n", tuple_id);
   ast_str_append(&body, 0, "<status><basic>%s</basic></status>\n", state == CC_OPEN ? "open" : "closed");
   ast_str_append(&body, 0, "</tuple>\n");
   ast_str_append(&body, 0, "</presence>\n");
   ast_copy_string(pidf_body, ast_str_buffer(body), size);
   return 0;
}

static int copy_all_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy all headers from one request to another.

Definition at line 9127 of file chan_sip.c.

References __get_header(), add_header(), and ast_strlen_zero().

Referenced by respprep().

{
   int start = 0;
   int copied = 0;
   for (;;) {
      const char *tmp = __get_header(orig, field, &start);

      if (ast_strlen_zero(tmp))
         break;
      /* Add what we're responding to */
      add_header(req, field, tmp);
      copied++;
   }
   return copied ? 0 : -1;
}

static int copy_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy one header field from one request to another.

Definition at line 9116 of file chan_sip.c.

References add_header(), ast_log(), ast_strlen_zero(), get_header(), and LOG_NOTICE.

Referenced by reqprep(), and respprep().

{
   const char *tmp = get_header(orig, field);

   if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
      return add_header(req, field, tmp);
   ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
   return -1;
}

static void copy_request	(	struct sip_request *	dst,
		const struct sip_request *	src
	)			[static]

copy SIP request (mostly used to save request for responses)

Definition at line 10792 of file chan_sip.c.

References ast_str_copy_string(), ast_str_create(), and ast_str_strlen().

Referenced by _sip_tcp_helper_thread(), handle_request_bye(), handle_request_invite(), handle_request_options(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), initialize_initreq(), parse_copy(), queue_request(), sip_park(), and sip_park_thread().

{
   /* XXX this function can encounter memory allocation errors, perhaps it
    * should return a value */

   struct ast_str *duplicate = dst->data;
   struct ast_str *duplicate_content = dst->content;

   /* copy the entire request then restore the original data and content
    * members from the dst request */
   memcpy(dst, src, sizeof(*dst));
   dst->data = duplicate;
   dst->content = duplicate_content;

   /* copy the data into the dst request */
   if (!dst->data && !(dst->data = ast_str_create(ast_str_strlen(src->data) + 1)))
      return;
   ast_str_copy_string(&dst->data, src->data);

   /* copy the content into the dst request (if it exists) */
   if (src->content) {
      if (!dst->content && !(dst->content = ast_str_create(ast_str_strlen(src->content) + 1)))
         return;
      ast_str_copy_string(&dst->content, src->content);
   }
}

static void copy_socket_data	(	struct sip_socket *	to_sock,
		const struct sip_socket *	from_sock
	)			[static]

Definition at line 4731 of file chan_sip.c.

References ao2_ref.

Referenced by create_addr_from_peer(), handle_request_do(), parse_register_contact(), sip_poke_peer(), and transmit_response_using_temp().

{
   if (to_sock->tcptls_session) {
      ao2_ref(to_sock->tcptls_session, -1);
      to_sock->tcptls_session = NULL;
   }

   if (from_sock->tcptls_session) {
      ao2_ref(from_sock->tcptls_session, +1);
   }

   *to_sock = *from_sock;
}

static struct ast_variable * copy_vars

(

struct ast_variable *

src

)

[static, read]

duplicate a list of channel variables,

Returns:

the copy.

Definition at line 2280 of file chan_sip.c.

References ast_variable_new(), and ast_variable::next.

{
   struct ast_variable *res = NULL, *tmp, *v = NULL;

   for (v = src ; v ; v = v->next) {
      if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
         tmp->next = res;
         res = tmp;
      }
   }
   return res;
}

static int copy_via_headers	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)			[static]

Copy SIP VIA Headers from the request to the response.

Note:

If the client indicates that it wishes to know the port we received from, it adds ;rport without an argument to the topmost via header. We need to add the port number (from our point of view) to that parameter.

	We always add ;received=<ip address> to the topmost via header.

Received: RFC 3261, rport RFC 3581

Definition at line 9151 of file chan_sip.c.

References __get_header(), add_header(), ast_copy_string(), ast_log(), ast_sockaddr_port, ast_sockaddr_stringify_addr(), ast_strlen_zero(), ast_test_flag, and LOG_NOTICE.

Referenced by respprep().

{
   int copied = 0;
   int start = 0;

   for (;;) {
      char new[512];
      const char *oh = __get_header(orig, field, &start);

      if (ast_strlen_zero(oh))
         break;

      if (!copied) { /* Only check for empty rport in topmost via header */
         char leftmost[512], *others, *rport;

         /* Only work on leftmost value */
         ast_copy_string(leftmost, oh, sizeof(leftmost));
         others = strchr(leftmost, ',');
         if (others)
             *others++ = '\0';

         /* Find ;rport;  (empty request) */
         rport = strstr(leftmost, ";rport");
         if (rport && *(rport+6) == '=')
            rport = NULL;     /* We already have a parameter to rport */

         if (((ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) || (rport && ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)))) {
            /* We need to add received port - rport */
            char *end;

            rport = strstr(leftmost, ";rport");

            if (rport) {
               end = strchr(rport + 1, ';');
               if (end)
                  memmove(rport, end, strlen(end) + 1);
               else
                  *rport = '\0';
            }

            /* Add rport to first VIA header if requested */
            snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
               leftmost, ast_sockaddr_stringify_addr(&p->recv),
               ast_sockaddr_port(&p->recv),
               others ? "," : "", others ? others : "");
         } else {
            /* We should *always* add a received to the topmost via */
            snprintf(new, sizeof(new), "%s;received=%s%s%s",
               leftmost, ast_sockaddr_stringify_addr(&p->recv),
               others ? "," : "", others ? others : "");
         }
         oh = new;   /* the header to copy */
      }  /* else add the following via headers untouched */
      add_header(req, field, oh);
      copied++;
   }
   if (!copied) {
      ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
      return -1;
   }
   return 0;
}

static int create_addr	(	struct sip_pvt *	dialog,
		const char *	opeer,
		struct ast_sockaddr *	addr,
		int	newdialog,
		struct ast_sockaddr *	remote_address
	)			[static]

create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success

Todo:

Fix this function. When we ask for SRV, we should check all transports In the future, we should first check NAPTR to find out transport preference

Definition at line 4949 of file chan_sip.c.

References AST_APP_ARG, AST_DECLARE_APP_ARGS, ast_get_srv(), ast_log(), AST_NONSTANDARD_RAW_ARGS, ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_strdupa, ast_string_field_set, bindaddr, create_addr_from_peer(), dialog_initialize_rtp(), FALSE, find_peer(), get_srv_protocol(), get_srv_service(), LOG_WARNING, MAXHOSTNAMELEN, obproxy_get(), ref_peer(), ref_proxy(), service, set_socket_transport(), sip_cfg, TRUE, and unref_peer().

{
   struct sip_peer *peer;
   char *peername, *peername2, *hostn;
   char host[MAXHOSTNAMELEN];
   char service[MAXHOSTNAMELEN];
   int srv_ret = 0;
   int tportno;

   AST_DECLARE_APP_ARGS(hostport,
      AST_APP_ARG(host);
      AST_APP_ARG(port);
   );

   peername = ast_strdupa(opeer);
   peername2 = ast_strdupa(opeer);
   AST_NONSTANDARD_RAW_ARGS(hostport, peername2, ':');

   if (hostport.port)
      dialog->portinuri = 1;

   dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
   dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
   peer = find_peer(peername, NULL, TRUE, FINDPEERS, FALSE, 0);

   if (peer) {
      int res;
      if (newdialog) {
         set_socket_transport(&dialog->socket, 0);
      }
      res = create_addr_from_peer(dialog, peer);
      if (!ast_sockaddr_isnull(remote_address)) {
         ast_sockaddr_copy(&dialog->sa, remote_address);
      }
      dialog->relatedpeer = ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer");
      unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
      return res;
   }

   if (dialog_initialize_rtp(dialog)) {
      return -1;
   }

   ast_string_field_set(dialog, tohost, hostport.host);
   dialog->allowed_methods &= ~sip_cfg.disallowed_methods;

   /* Get the outbound proxy information */
   ref_proxy(dialog, obproxy_get(dialog, NULL));

   if (addr) {
      /* This address should be updated using dnsmgr */
      ast_sockaddr_copy(&dialog->sa, addr);
   } else {

      /* Let's see if we can find the host in DNS. First try DNS SRV records,
         then hostname lookup */
      /*! \todo Fix this function. When we ask for SRV, we should check all transports
           In the future, we should first check NAPTR to find out transport preference
       */
      hostn = peername;
      /* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
       * an A record lookup should be used instead of SRV.
       */
      if (!hostport.port && sip_cfg.srvlookup) {
         snprintf(service, sizeof(service), "_%s._%s.%s", 
             get_srv_service(dialog->socket.type),
             get_srv_protocol(dialog->socket.type), peername);
         if ((srv_ret = ast_get_srv(NULL, host, sizeof(host), &tportno,
                     service)) > 0) {
            hostn = host;
         }
      }

      if (ast_sockaddr_resolve_first(&dialog->sa, hostn, 0)) {
         ast_log(LOG_WARNING, "No such host: %s\n", peername);
      }

      if (srv_ret > 0) {
         ast_sockaddr_set_port(&dialog->sa, tportno);
      }
   }

   if (!dialog->socket.type)
      set_socket_transport(&dialog->socket, SIP_TRANSPORT_UDP);
   if (!dialog->socket.port) {
      dialog->socket.port = htons(ast_sockaddr_port(&bindaddr));
   }

   if (!ast_sockaddr_port(&dialog->sa)) {
      ast_sockaddr_set_port(&dialog->sa,
                  (dialog->socket.type == SIP_TRANSPORT_TLS) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }
   ast_sockaddr_copy(&dialog->recv, &dialog->sa);
   return 0;
}

static int create_addr_from_peer	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)			[static]

Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.

Returns:

-1 on error, 0 on success.

Definition at line 4803 of file chan_sip.c.

References accountcode, ao2_t_link, ao2_t_unlink, ast_cc_copy_config_params(), ast_clear_flag, ast_copy_flags, ast_debug, ast_duplicate_ha_list(), ast_rtp_codecs_packetization_set(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_instance_set_hold_timeout(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_timeout(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify_host(), ast_strdupa, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_udptl_destroy(), ast_udptl_new_with_bindaddr(), bindaddr, check_request_transport, cid_name, cid_num, context, copy_socket_data(), copy_vars(), dialog_initialize_rtp(), language, mohinterpret, mohsuggest, obproxy_get(), parkinglot, ref_proxy(), and set_t38_capabilities().

Referenced by create_addr(), and sip_send_mwi_to_peer().

{
   /* this checks that the dialog is contacting the peer on a valid
    * transport type based on the peers transport configuration,
    * otherwise, this function bails out */
   if (dialog->socket.type && check_request_transport(peer, dialog))
      return -1;
   copy_socket_data(&dialog->socket, &peer->socket);

   if (!(ast_sockaddr_isnull(&peer->addr) && ast_sockaddr_isnull(&peer->defaddr)) &&
       (!peer->maxms || ((peer->lastms >= 0)  && (peer->lastms <= peer->maxms)))) {
      dialog->sa = ast_sockaddr_isnull(&peer->addr) ? peer->defaddr : peer->addr;
      dialog->recv = dialog->sa;
   } else
      return -1;

   /* XXX TODO: get flags directly from peer only as they are needed using dialog->relatedpeer */
   ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
   dialog->capability = peer->capability;
   dialog->prefs = peer->prefs;
   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
      /* t38pt_udptl was enabled in the peer and not in [general] */
      if (dialog->udptl || (!dialog->udptl && (dialog->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr)))) {
         dialog->t38_maxdatagram = peer->t38_maxdatagram;
         set_t38_capabilities(dialog);
      } else {
         /* It is impossible to support T38 without udptl */
         ast_debug(1, "UDPTL creation failed on dialog.\n");
         ast_clear_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT);
      }
   } else if (dialog->udptl) {
      ast_udptl_destroy(dialog->udptl);
      dialog->udptl = NULL;
   }

   ast_string_field_set(dialog, engine, peer->engine);

   if (dialog_initialize_rtp(dialog)) {
      return -1;
   }

   if (dialog->rtp) { /* Audio */
      ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
      ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
      /* Set Frame packetization */
      ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
      dialog->autoframing = peer->autoframing;
   }
   if (dialog->vrtp) { /* Video */
      ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
   }
   if (dialog->trtp) { /* Realtime text */
      ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
   }

   /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
   ast_string_field_set(dialog, peername, peer->name);
   ast_string_field_set(dialog, authname, peer->username);
   ast_string_field_set(dialog, username, peer->username);
   ast_string_field_set(dialog, peersecret, peer->secret);
   ast_string_field_set(dialog, peermd5secret, peer->md5secret);
   ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
   ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
   ast_string_field_set(dialog, tohost, peer->tohost);
   ast_string_field_set(dialog, fullcontact, peer->fullcontact);
   ast_string_field_set(dialog, accountcode, peer->accountcode);
   ast_string_field_set(dialog, context, peer->context);
   ast_string_field_set(dialog, cid_num, peer->cid_num);
   ast_string_field_set(dialog, cid_name, peer->cid_name);
   ast_string_field_set(dialog, cid_tag, peer->cid_tag);
   ast_string_field_set(dialog, mwi_from, peer->mwi_from);
   ast_string_field_set(dialog, parkinglot, peer->parkinglot);
   ast_string_field_set(dialog, engine, peer->engine);
   ref_proxy(dialog, obproxy_get(dialog, peer));
   dialog->callgroup = peer->callgroup;
   dialog->pickupgroup = peer->pickupgroup;
   dialog->allowtransfer = peer->allowtransfer;
   dialog->jointnoncodeccapability = dialog->noncodeccapability;
   dialog->rtptimeout = peer->rtptimeout;
   dialog->peerauth = peer->auth;
   dialog->maxcallbitrate = peer->maxcallbitrate;
   dialog->disallowed_methods = peer->disallowed_methods;
   ast_cc_copy_config_params(dialog->cc_params, peer->cc_params);
   if (ast_strlen_zero(dialog->tohost))
      ast_string_field_set(dialog, tohost, ast_sockaddr_stringify_host(&dialog->sa));
   if (!ast_strlen_zero(peer->fromdomain)) {
      ast_string_field_set(dialog, fromdomain, peer->fromdomain);
      if (!dialog->initreq.headers) {
         char *c;
         char *tmpcall = ast_strdupa(dialog->callid);
         /* this sure looks to me like we are going to change the callid on this dialog!! */
         c = strchr(tmpcall, '@');
         if (c) {
            *c = '\0';
            ao2_t_unlink(dialogs, dialog, "About to change the callid -- remove the old name");
            ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
            ao2_t_link(dialogs, dialog, "New dialog callid -- inserted back into table");
         }
      }
   }
   if (!ast_strlen_zero(peer->fromuser))
      ast_string_field_set(dialog, fromuser, peer->fromuser);
   if (!ast_strlen_zero(peer->language))
      ast_string_field_set(dialog, language, peer->language);
   /* Set timer T1 to RTT for this peer (if known by qualify=) */
   /* Minimum is settable or default to 100 ms */
   /* If there is a maxms and lastms from a qualify use that over a manual T1
      value. Otherwise, use the peer's T1 value. */
   if (peer->maxms && peer->lastms)
      dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      dialog->timer_t1 = peer->timer_t1;

   /* Set timer B to control transaction timeouts, the peer setting is the default and overrides
      the known timer */
   if (peer->timer_b)
      dialog->timer_b = peer->timer_b;
   else
      dialog->timer_b = 64 * dialog->timer_t1;

   if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
      dialog->noncodeccapability |= AST_RTP_DTMF;
   else
      dialog->noncodeccapability &= ~AST_RTP_DTMF;
   dialog->directmediaha = ast_duplicate_ha_list(peer->directmediaha);
   if (peer->call_limit)
      ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
   if (!dialog->portinuri)
      dialog->portinuri = peer->portinuri;
   dialog->chanvars = copy_vars(peer->chanvars);
   if (peer->fromdomainport)
      dialog->fromdomainport = peer->fromdomainport;

   return 0;
}

static struct sip_epa_entry* create_epa_entry	(	const char *const	event_package,
		const char *const	destination
	)			[static, read]

Definition at line 900 of file chan_sip.c.

References ao2_t_alloc, ast_copy_string(), and find_static_data().

Referenced by sip_cc_monitor_suspend().

{
   struct sip_epa_entry *epa_entry;
   const struct epa_static_data *static_data;

   if (!(static_data = find_static_data(event_package))) {
      return NULL;
   }

   if (!(epa_entry = ao2_t_alloc(sizeof(*epa_entry), static_data->destructor, "Allocate new EPA entry"))) {
      return NULL;
   }

   epa_entry->static_data = static_data;
   ast_copy_string(epa_entry->destination, destination, sizeof(epa_entry->destination));
   return epa_entry;
}

static struct sip_esc_entry* create_esc_entry	(	struct event_state_compositor *	esc,
		struct sip_request *	req,
		const int	expires
	)			[static, read]

Definition at line 1026 of file chan_sip.c.

References ao2_alloc, ao2_ref, ast_sched_add(), create_new_sip_etag(), esc_entry_destructor(), event_state_compositor::name, and publish_expire().

Referenced by handle_sip_publish_initial().

{
   struct sip_esc_entry *esc_entry;
   int expires_ms;

   if (!(esc_entry = ao2_alloc(sizeof(*esc_entry), esc_entry_destructor))) {
      return NULL;
   }

   esc_entry->event = esc->name;

   expires_ms = expires * 1000;
   /* Bump refcount for scheduler */
   ao2_ref(esc_entry, +1);
   esc_entry->sched_id = ast_sched_add(sched, expires_ms, publish_expire, esc_entry);

   /* Note: This links the esc_entry into the ESC properly */
   create_new_sip_etag(esc_entry, 0);

   return esc_entry;
}

static void create_new_sip_etag	(	struct sip_esc_entry *	esc_entry,
		int	is_linked
	)			[static]

Definition at line 1013 of file chan_sip.c.

References ao2_link, ao2_unlink, ast_assert, ast_atomic_fetchadd_int(), event_state_compositor::compositor, and get_esc().

Referenced by create_esc_entry(), and transmit_response_with_sip_etag().

{
   int new_etag = ast_atomic_fetchadd_int(&esc_etag_counter, +1);
   struct event_state_compositor *esc = get_esc(esc_entry->event);

   ast_assert(esc != NULL);
   if (is_linked) {
      ao2_unlink(esc->compositor, esc_entry);
   }
   snprintf(esc_entry->entity_tag, sizeof(esc_entry->entity_tag), "%d", new_etag);
   ao2_link(esc->compositor, esc_entry);
}

static void deinit_req

(

struct sip_request *

req

)

[static]

Deinitialize SIP response/request.

Definition at line 9370 of file chan_sip.c.

References ast_free.

Referenced by __sip_destroy(), _sip_tcp_helper_thread(), send_request(), send_response(), sip_park(), sip_park_thread(), and sipsock_read().

{
   if (req->data) {
      ast_free(req->data);
      req->data = NULL;
   }
   if (req->content) {
      ast_free(req->content);
      req->content = NULL;
   }
}

static void destroy_association

(

struct sip_peer *

peer

)

[static]

Remove registration data from realtime database or AST/DB when registration expires.

Definition at line 12664 of file chan_sip.c.

References ast_check_realtime(), ast_db_del(), ast_update_realtime(), SENTINEL, and sip_cfg.

Referenced by build_peer(), expire_register(), and parse_register_contact().

{
   int realtimeregs = ast_check_realtime("sipregs");
   char *tablename = (realtimeregs) ? "sipregs" : "sippeers";

   if (!sip_cfg.ignore_regexpire) {
      if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
         ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", peer->deprecated_username ? "username" : "defaultuser", "", "regserver", "", "useragent", "", "lastms", "", SENTINEL);
      } else {
         ast_db_del("SIP/Registry", peer->name);
         ast_db_del("SIP/PeerMethods", peer->name);
      }
   }
}

static void destroy_escs

(

void

)

[static]

Definition at line 1060 of file chan_sip.c.

References ao2_ref, ARRAY_LEN, and event_state_compositors.

Referenced by unload_module().

{
   int i;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      ao2_ref(event_state_compositors[i].compositor, -1);
   }
}

static void destroy_mailbox

(

struct sip_mailbox *

mailbox

)

[static]

Destroy mailbox subscriptions

Definition at line 4259 of file chan_sip.c.

References ast_event_unsubscribe(), and ast_free.

Referenced by build_peer(), and clear_peer_mailboxes().

{
   if (mailbox->event_sub)
      ast_event_unsubscribe(mailbox->event_sub);
   ast_free(mailbox);
}

static int determine_firstline_parts

(

struct sip_request *

req

)

[static]

Parse first line of incoming SIP request.

Definition at line 10889 of file chan_sip.c.

References ast_debug, ast_skip_blanks(), ast_skip_nonblanks(), and ast_trim_blanks().

Referenced by parse_request().

{
   char *e = ast_skip_blanks(req->data->str);   /* there shouldn't be any */
   char *local_rlPart1;

   if (!*e)
      return -1;
   req->rlPart1 = e - req->data->str;  /* method or protocol */
   local_rlPart1 = e;
   e = ast_skip_nonblanks(e);
   if (*e)
      *e++ = '\0';
   /* Get URI or status code */
   e = ast_skip_blanks(e);
   if ( !*e )
      return -1;
   ast_trim_blanks(e);

   if (!strcasecmp(local_rlPart1, "SIP/2.0") ) { /* We have a response */
      if (strlen(e) < 3)   /* status code is 3 digits */
         return -1;
      req->rlPart2 = e - req->data->str;
   } else { /* We have a request */
      if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */
         ast_debug(3, "Oops. Bogus uri in <> %s\n", e);
         e++;
         if (!*e)
            return -1;
      }
      req->rlPart2 = e - req->data->str;  /* URI */
      e = ast_skip_nonblanks(e);
      if (*e)
         *e++ = '\0';
      e = ast_skip_blanks(e);
      if (strcasecmp(e, "SIP/2.0") ) {
         ast_debug(3, "Skipping packet - Bad request protocol %s\n", e);
         return -1;
      }
   }
   return 1;
}

static enum sip_publish_type determine_sip_publish_type	(	struct sip_request *	req,
		const char *const	event,
		const char *const	etag,
		const char *const	expires,
		int *	expires_int
	)			[static]

Definition at line 22393 of file chan_sip.c.

References ast_assert, and ast_strlen_zero().

Referenced by handle_request_publish().

{
   int etag_present = !ast_strlen_zero(etag);
   int body_present = req->lines > 0;

   ast_assert(expires_int != NULL);

   if (ast_strlen_zero(expires)) {
      /* Section 6, item 4, second bullet point of RFC 3903 says to
       * use a locally-configured default expiration if none is provided
       * in the request
       */
      *expires_int = DEFAULT_PUBLISH_EXPIRES;
   } else if (sscanf(expires, "%30d", expires_int) != 1) {
      return SIP_PUBLISH_UNKNOWN;
   }

   if (*expires_int == 0) {
      return SIP_PUBLISH_REMOVE;
   } else if (!etag_present && body_present) {
      return SIP_PUBLISH_INITIAL;
   } else if (etag_present && !body_present) {
      return SIP_PUBLISH_REFRESH;
   } else if (etag_present && body_present) {
      return SIP_PUBLISH_MODIFY;
   }

   return SIP_PUBLISH_UNKNOWN;
}

static int dialog_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

The only member of the dialog used here callid string

Definition at line 28085 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_pvt *pvt = obj, *pvt2 = arg;

   return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static int dialog_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 15699 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and ast_cli_args::fd.

Referenced by sip_show_objects().

{
   struct sip_pvt *pvt = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   struct ast_cli_args *a = (struct ast_cli_args *) arg;
   
   ast_cli(a->fd, "name: %s\ntype: dialog\nobjflags: %d\nrefcount: %d\n\n",
      pvt->callid, 0, refc);
   return 0;
}

static int dialog_find_multiple	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

Same as dialog_cmp_cb, except without the CMP_STOP on match

Definition at line 28075 of file chan_sip.c.

References CMP_MATCH.

Referenced by find_call().

{
   struct sip_pvt *pvt = obj, *pvt2 = arg;

   return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH : 0;
}

static int dialog_hash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Note:

The only member of the dialog used here callid string

Definition at line 28065 of file chan_sip.c.

References ast_str_case_hash().

Referenced by load_module().

{
   const struct sip_pvt *pvt = obj;

   return ast_str_case_hash(pvt->callid);
}

static int dialog_initialize_rtp

(

struct sip_pvt *

dialog

)

[static]

Initialize RTP portion of a dialog.

Returns:

-1 on failure, 0 on success

Definition at line 4748 of file chan_sip.c.

References AST_FORMAT_VIDEO_MASK, ast_rtp_instance_new(), ast_rtp_instance_set_hold_timeout(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_qos(), ast_rtp_instance_set_timeout(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, AST_RTP_PROPERTY_RTCP, ast_sockaddr_copy(), ast_test_flag, bindaddr, do_setnat(), cfsip_methods::need_rtp, and sip_methods.

Referenced by check_peer_ok(), check_user_full(), create_addr(), and create_addr_from_peer().

{
   struct ast_sockaddr bindaddr_tmp;

   if (!sip_methods[dialog->method].need_rtp) {
      return 0;
   }

   ast_sockaddr_copy(&bindaddr_tmp, &bindaddr);
   if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
      return -1;
   }

   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) ||
         (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (dialog->capability & AST_FORMAT_VIDEO_MASK))) {
      if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
         return -1;
      }
      ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);

      ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
   }

   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
      if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
         return -1;
      }
      ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);

      ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
   }

   ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
   ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);

   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));

   ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, global_cos_audio, "SIP RTP");

   do_setnat(dialog);

   return 0;
}

static int dialog_needdestroy	(	void *	dialogobj,
		void *	arg,
		int	flags
	)			[static]

Match dialogs that need to be destroyed.

This is used with ao2_callback to unlink/delete all dialogs that are marked needdestroy. It will return CMP_MATCH for candidates, and they will be unlinked.

Todo:

Re-work this to improve efficiency. Currently, this function is called on _every_ dialog after processing _every_ incoming SIP/UDP packet, or potentially even more often when the scheduler has entries to run.

Definition at line 15818 of file chan_sip.c.

References ast_debug, ast_rtp_instance_get_bridged(), check_rtp_timeout(), dialog_unlink_all(), FALSE, sip_methods, sip_pvt_trylock, sip_pvt_unlock, cfsip_methods::text, and TRUE.

Referenced by do_monitor().

{
   struct sip_pvt *dialog = dialogobj;
   time_t *t = arg;

   if (sip_pvt_trylock(dialog)) {
      /* Don't block the monitor thread.  This function is called often enough
       * that we can wait for the next time around. */
      return 0;
   }

   /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
   if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
      ast_debug(2, "Bridge still active.  Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
      sip_pvt_unlock(dialog);
      return 0;
   }

   if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
      ast_debug(2, "Bridge still active.  Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
      sip_pvt_unlock(dialog);
      return 0;
   }

   /* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
   check_rtp_timeout(dialog, *t);

   /* If we have sessions that needs to be destroyed, do it now */
   /* Check if we have outstanding requests not responsed to or an active call
      - if that's the case, wait with destruction */
   if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
      /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
      if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
         ast_debug(2, "Bridge still active.  Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }
      
      if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
         ast_debug(2, "Bridge still active.  Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }

      sip_pvt_unlock(dialog);
      /* no, the unlink should handle this: dialog_unref(dialog, "needdestroy: one more refcount decrement to allow dialog to be destroyed"); */
      /* the CMP_MATCH will unlink this dialog from the dialog hash table */
      dialog_unlink_all(dialog, TRUE, FALSE);
      return 0; /* the unlink_all should unlink this from the table, so.... no need to return a match */
   }

   sip_pvt_unlock(dialog);

   return 0;
}

struct sip_pvt* dialog_ref_debug	(	struct sip_pvt *	p,
		char *	tag,
		char *	file,
		int	line,
		const char *	func
	)			[read]

Definition at line 2172 of file chan_sip.c.

References __ao2_ref_debug(), ao2_ref, ast_log(), and LOG_ERROR.

{
   if (p)
#ifdef REF_DEBUG
      __ao2_ref_debug(p, 1, tag, file, line, func);
#else
      ao2_ref(p, 1);
#endif
   else
      ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
   return p;
}

void* dialog_unlink_all	(	struct sip_pvt *	dialog,
		int	lockowner,
		int	lockdialoglist
	)

Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.

Note:

A reference to the dialog must be held before calling this function, and this function does not release that reference.

Definition at line 2702 of file chan_sip.c.

References ao2_t_unlink, ast_channel_lock, ast_channel_unlock, ast_debug, ast_extension_state_del(), ast_free, AST_SCHED_DEL, AST_SCHED_DEL_UNREF, and registry_unref().

Referenced by __sip_autodestruct(), __sip_subscribe_mwi_do(), dialog_needdestroy(), handle_request_subscribe(), manager_sipnotify(), reload_config(), sip_cli_notify(), sip_destroy_peer(), sip_poke_noanswer(), sip_poke_peer(), sip_registry_destroy(), sip_request_call(), sip_send_mwi_to_peer(), transmit_publish(), transmit_register(), and unload_module().

{
   struct sip_pkt *cp;

   dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");

   ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");

   /* Unlink us from the owner (channel) if we have one */
   if (dialog->owner) {
      if (lockowner)
         ast_channel_lock(dialog->owner);
      ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
      dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
      if (lockowner) {
         ast_channel_unlock(dialog->owner);
      }
   }
   if (dialog->registry) {
      if (dialog->registry->call == dialog) {
         dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
      }
      dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
   }
   if (dialog->stateid > -1) {
      ast_extension_state_del(dialog->stateid, NULL);
      dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
      dialog->stateid = -1; /* shouldn't we 'zero' this out? */
   }
   /* Remove link from peer to subscription of MWI */
   if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog) {
      dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   }
   if (dialog->relatedpeer && dialog->relatedpeer->call == dialog) {
      dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   }

   /* remove all current packets in this dialog */
   while((cp = dialog->packets)) {
      dialog->packets = dialog->packets->next;
      AST_SCHED_DEL(sched, cp->retransid);
      dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
      if (cp->data) {
         ast_free(cp->data);
      }
      ast_free(cp);
   }

   AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));

   AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
   
   if (dialog->autokillid > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));
   }

   if (dialog->request_queue_sched_id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id, dialog_unref(dialog, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
   }

   AST_SCHED_DEL_UNREF(sched, dialog->provisional_keepalive_sched_id, dialog_unref(dialog, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));

   if (dialog->t38id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->t38id, dialog_unref(dialog, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
   }

   dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
   return NULL;
}

struct sip_pvt* dialog_unref_debug	(	struct sip_pvt *	p,
		char *	tag,
		char *	file,
		int	line,
		const char *	func
	)			[read]

Definition at line 2185 of file chan_sip.c.

References __ao2_ref_debug(), and ao2_ref.

{
   if (p)
#ifdef REF_DEBUG
      __ao2_ref_debug(p, -1, tag, file, line, func);
#else
      ao2_ref(p, -1);
#endif
   return NULL;
}

static void disable_dsp_detect

(

struct sip_pvt *

p

)

[static]

Definition at line 3971 of file chan_sip.c.

References ast_dsp_free().

Referenced by sip_dtmfmode(), sip_hangup(), and sip_setoption().

{
   if (p->dsp) {
      ast_dsp_free(p->dsp);
      p->dsp = NULL;
   }
}

static int do_magic_pickup	(	struct ast_channel *	channel,
		const char *	extension,
		const char *	context
	)			[static]

Note:

No channel or pvt locks should be held while calling this function.

Definition at line 20697 of file chan_sip.c.

References ast_debug, ast_log(), AST_MAX_CONTEXT, AST_MAX_EXTENSION, ast_str_alloca, ast_str_set(), LOG_ERROR, pbx_exec(), pbx_findapp(), sip_cfg, and str.

Referenced by handle_request_invite().

{
   struct ast_str *str = ast_str_alloca(AST_MAX_EXTENSION + AST_MAX_CONTEXT + 2);
   struct ast_app *pickup = pbx_findapp("Pickup");

   if (!pickup) {
      ast_log(LOG_ERROR, "Unable to perform pickup: Application 'Pickup' not loaded (app_directed_pickup.so).\n");
      return -1;
   }

   ast_str_set(&str, 0, "%s@%s", extension, sip_cfg.notifycid == IGNORE_CONTEXT ? "PICKUPMARK" : context);

   ast_debug(2, "About to call Pickup(%s)\n", str->str);

   /* There is no point in capturing the return value since pickup_exec
      doesn't return anything meaningful unless the passed data is an empty
      string (which in our case it will not be) */
   pbx_exec(channel, pickup, str->str);

   return 0;
}

static void * do_monitor

(

void *

data

)

[static]

The SIP monitoring thread.

Note:

This thread monitors all the SIP sessions and peers that needs notification of mwi (and thus do not have a separate thread) indefinitely

Definition at line 24231 of file chan_sip.c.

References ao2_t_callback, ast_debug, ast_io_add(), ast_io_change(), AST_IO_IN, ast_io_remove(), ast_io_wait(), ast_mutex_lock, ast_mutex_unlock, ast_sched_runq(), ast_sched_wait(), ast_verb, dialog_needdestroy(), FALSE, monlock, OBJ_MULTIPLE, OBJ_NODATA, OBJ_UNLINK, sip_do_reload(), sip_reload_lock, and sipsock_read().

{
   int res;
   time_t t;
   int reloading;

   /* Add an I/O event to our SIP UDP socket */
   if (sipsock > -1)
      sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);

   /* From here on out, we die whenever asked */
   for(;;) {
      /* Check for a reload request */
      ast_mutex_lock(&sip_reload_lock);
      reloading = sip_reloading;
      sip_reloading = FALSE;
      ast_mutex_unlock(&sip_reload_lock);
      if (reloading) {
         ast_verb(1, "Reloading SIP\n");
         sip_do_reload(sip_reloadreason);

         /* Change the I/O fd of our UDP socket */
         if (sipsock > -1) {
            if (sipsock_read_id)
               sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
            else
               sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
         } else if (sipsock_read_id) {
            ast_io_remove(io, sipsock_read_id);
            sipsock_read_id = NULL;
         }
      }

      /* Check for dialogs needing to be killed */
      t = time(NULL);
      /* don't scan the dialogs list if it hasn't been a reasonable period
         of time since the last time we did it (when MWI is being sent, we can
         get back to this point every millisecond or less)
      */
      ao2_t_callback(dialogs, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy, &t,
            "callback to remove dialogs w/needdestroy");

      /* the old methodology would be to restart the search for dialogs to delete with every
         dialog that was found and destroyed, probably because the list contents would change,
         so we'd need to restart. This isn't the best thing to do with callbacks. */

      /* XXX TODO The scheduler usage in this module does not have sufficient
       * synchronization being done between running the scheduler and places
       * scheduling tasks.  As it is written, any scheduled item may not run
       * any sooner than about  1 second, regardless of whether a sooner time
       * was asked for. */

      pthread_testcancel();
      /* Wait for sched or io */
      res = ast_sched_wait(sched);
      if ((res < 0) || (res > 1000))
         res = 1000;
      res = ast_io_wait(io, res);
      if (res > 20)
         ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res);
      ast_mutex_lock(&monlock);
      res = ast_sched_runq(sched);
      if (res >= 20)
         ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res);
      ast_mutex_unlock(&monlock);
   }

   /* Never reached */
   return NULL;
}

static int do_proxy_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code,
		int	sipmethod,
		int	init
	)			[static]

Add authentication on outbound SIP packet.

Definition at line 17917 of file chan_sip.c.

References ast_calloc, ast_debug, auth_headers(), reply_digest(), sip_methods, cfsip_methods::text, and transmit_invite().

Referenced by handle_response(), handle_response_invite(), handle_response_notify(), handle_response_publish(), handle_response_refer(), handle_response_subscribe(), and handle_response_update().

{
   char *header, *respheader;
   char digest[1024];

   if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options))))
      return -2;

   p->authtries++;
   auth_headers(code, &header, &respheader);
   ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) {
      /* No way to authenticate */
      return -1;
   }
   /* Now we have a reply digest */
   p->options->auth = digest;
   p->options->authheader = respheader;
   return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init, NULL);
}

static int do_register_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code
	)			[static]

Authenticate for outbound registration.

Definition at line 17893 of file chan_sip.c.

References append_history, ast_verbose, auth_headers(), reply_digest(), sip_debug_test_pvt(), and transmit_register().

Referenced by handle_response_register().

{
   char *header, *respheader;
   char digest[1024];

   p->authtries++;
   auth_headers(code, &header, &respheader);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
      /* There's nothing to use for authentication */
      /* No digest challenge in request */
      if (sip_debug_test_pvt(p) && p->registry)
         ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
         /* No old challenge */
      return -1;
   }
   if (p->do_history)
      append_history(p, "RegistryAuth", "Try: %d", p->authtries);
   if (sip_debug_test_pvt(p) && p->registry)
      ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
   return transmit_register(p->registry, SIP_REGISTER, digest, respheader);
}

static void do_setnat

(

struct sip_pvt *

p

)

[static]

Set nat mode on the various data sockets.

Definition at line 4642 of file chan_sip.c.

References ast_debug, ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_NAT, ast_test_flag, and ast_udptl_setnat().

Referenced by check_peer_ok(), dialog_initialize_rtp(), sip_alloc(), and transmit_response_using_temp().

{
   const char *mode;
   int natflags;

   natflags = ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
   mode = natflags ? "On" : "Off";

   if (p->rtp) {
      ast_debug(1, "Setting NAT on RTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_NAT, natflags);
   }
   if (p->vrtp) {
      ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_NAT, natflags);
   }
   if (p->udptl) {
      ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
      ast_udptl_setnat(p->udptl, natflags);
   }
   if (p->trtp) {
      ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_NAT, natflags);
   }
}

static const char * domain_mode_to_text

(

const enum domain_mode

mode

)

[static]

Print domain mode to cli.

Definition at line 16027 of file chan_sip.c.

Referenced by sip_show_domains().

{
   switch (mode) {
   case SIP_DOMAIN_AUTO:
      return "[Automatic]";
   case SIP_DOMAIN_CONFIG:
      return "[Configured]";
   }

   return "";
}

static const char * dtmfmode2str

(

int

mode

)

[static]

Convert DTMF mode to printable string.

Definition at line 15755 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), sip_show_channel(), and sip_show_settings().

{
   return map_x_s(dtmfstr, mode, "<error>");
}

static void enable_dsp_detect

(

struct sip_pvt *

p

)

[static]

Definition at line 3938 of file chan_sip.c.

References ast_dsp_new(), ast_dsp_set_digitmode(), ast_dsp_set_features(), AST_RTP_DTMF_MODE_INBAND, ast_rtp_instance_dtmf_mode_set(), ast_test_flag, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_RELAXDTMF, DSP_FEATURE_DIGIT_DETECT, and DSP_FEATURE_FAX_DETECT.

Referenced by sip_dtmfmode(), sip_new(), and sip_setoption().

{
   int features = 0;

   if (p->dsp) {
      return;
   }

   if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
            (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      if (!p->rtp || ast_rtp_instance_dtmf_mode_set(p->rtp, AST_RTP_DTMF_MODE_INBAND)) {
         features |= DSP_FEATURE_DIGIT_DETECT;
                }
   }

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
      features |= DSP_FEATURE_FAX_DETECT;
   }

   if (!features) {
      return;
   }

   if (!(p->dsp = ast_dsp_new())) {
      return;
   }

   ast_dsp_set_features(p->dsp, features);
   if (global_relaxdtmf) {
      ast_dsp_set_digitmode(p->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
   }
}

static int esc_cmp_fn	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 972 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by initialize_escs().

{
   struct sip_esc_entry *entry1 = obj;
   struct sip_esc_entry *entry2 = arg;

   return (!strcmp(entry1->entity_tag, entry2->entity_tag)) ? (CMP_MATCH | CMP_STOP) : 0;
}

static void esc_entry_destructor

(

void *

obj

)

[static]

Definition at line 958 of file chan_sip.c.

References AST_SCHED_DEL.

Referenced by create_esc_entry().

{
   struct sip_esc_entry *esc_entry = obj;
   if (esc_entry->sched_id > -1) {
      AST_SCHED_DEL(sched, esc_entry->sched_id);
   }
}

static int esc_hash_fn	(	const void *	obj,
		const int	flags
	)			[static]

Definition at line 966 of file chan_sip.c.

References ast_str_hash().

Referenced by initialize_escs().

{
   const struct sip_esc_entry *entry = obj;
   return ast_str_hash(entry->entity_tag);
}

static int expire_register

(

const void *

data

)

[static]

Expire registration of SIP peer.

Definition at line 12693 of file chan_sip.c.

References ao2_ref, ast_debug, AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_test_flag, destroy_association(), EVENT_FLAG_SYSTEM, FALSE, manager_event, register_peer_exten(), set_socket_transport(), unlink_peer_from_tables(), and unref_peer().

Referenced by parse_register_contact(), realtime_peer(), reg_source_db(), sip_show_sched(), and sip_unregister().

{
   struct sip_peer *peer = (struct sip_peer *)data;

   if (!peer) {      /* Hmmm. We have no peer. Weird. */
      return 0;
   }

   peer->expire = -1;
   peer->portinuri = 0;

   destroy_association(peer); /* remove registration data from storage */
   set_socket_transport(&peer->socket, peer->default_outbound_transport);

   if (peer->socket.tcptls_session) {
      ao2_ref(peer->socket.tcptls_session, -1);
      peer->socket.tcptls_session = NULL;
   }

   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
   register_peer_exten(peer, FALSE);   /* Remove regexten */
   ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);

   /* Do we need to release this peer from memory?
      Only for realtime peers and autocreated peers
   */
   if (peer->is_realtime) {
      ast_debug(3, "-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs);
   }

   if (peer->selfdestruct ||
       ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
      unlink_peer_from_tables(peer);
   }

   /* Only clear the addr after we check for destruction.  The addr must remain
    * in order to unlink from the peers_by_ip container correctly */
   memset(&peer->addr, 0, sizeof(peer->addr));

   unref_peer(peer, "removing peer ref for expire_register");

   return 0;
}

static void extract_uri	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Check Contact: URI of SIP message.

Definition at line 10997 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, ast_strlen_zero(), get_header(), and remove_uri_parameters().

Referenced by handle_incoming(), and handle_request_invite().

{
   char stripped[SIPBUFSIZE];
   char *c;

   ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));
   c = get_in_brackets(stripped);
   /* Cut the URI at the at sign after the @, not in the username part */
   c = remove_uri_parameters(c);
   if (!ast_strlen_zero(c)) {
      ast_string_field_set(p, uri, c);
   }

}

static const char* faxec2str

(

int

faxec

)

[static]

Definition at line 16191 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), and sip_show_settings().

{
   return map_x_s(faxecmodes, faxec, "Unknown");
}

static int finalize_content

(

struct sip_request *

req

)

[static]

Add 'Content-Length' header and content to SIP message.

Definition at line 9083 of file chan_sip.c.

References add_header(), ast_log(), ast_str_append(), ast_str_buffer(), ast_str_strlen(), and LOG_WARNING.

Referenced by send_request(), and send_response().

{
   char clen[10];

   if (req->lines) {
      ast_log(LOG_WARNING, "finalize_content() called on a message that has already been finalized\n");
      return -1;
   }

   snprintf(clen, sizeof(clen), "%zd", ast_str_strlen(req->content));
   add_header(req, "Content-Length", clen);

   if (ast_str_strlen(req->content)) {
      ast_str_append(&req->data, 0, "\r\n%s", ast_str_buffer(req->content));
      req->len = ast_str_strlen(req->data);
   }
   req->lines = ast_str_strlen(req->content) ? 1 : 0;
   return 0;
}

static const char * find_alias	(	const char *	name,
		const char *	_default
	)			[static]

Find compressed SIP alias.

Structure for conversion between compressed SIP and "normal" SIP

Definition at line 6740 of file chan_sip.c.

References aliases, and ARRAY_LEN.

{
   /*! \brief Structure for conversion between compressed SIP and "normal" SIP */
   static const struct cfalias {
      char * const fullname;
      char * const shortname;
   } aliases[] = {
      { "Content-Type",  "c" },
      { "Content-Encoding",    "e" },
      { "From",       "f" },
      { "Call-ID",       "i" },
      { "Contact",       "m" },
      { "Content-Length",   "l" },
      { "Subject",       "s" },
      { "To",         "t" },
      { "Supported",     "k" },
      { "Refer-To",      "r" },
      { "Referred-By",   "b" },
      { "Allow-Events",  "u" },
      { "Event",      "o" },
      { "Via",     "v" },
      { "Accept-Contact",      "a" },
      { "Reject-Contact",      "j" },
      { "Request-Disposition", "d" },
      { "Session-Expires",     "x" },
      { "Identity",            "y" },
      { "Identity-Info",       "n" },
   };
   int x;

   for (x = 0; x < ARRAY_LEN(aliases); x++) {
      if (!strcasecmp(aliases[x].fullname, name))
         return aliases[x].shortname;
   }

   return _default;
}

static int find_by_callid_helper	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 1652 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and ast_cc_agent::private_data.

Referenced by find_sip_cc_agent_by_original_callid().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   struct sip_pvt *call_pvt = arg;

   return !strcmp(agent_pvt->original_callid, call_pvt->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static int find_by_name	(	void *	obj,
		void *	arg,
		void *	data,
		int	flags
	)			[static]

Definition at line 4553 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and match().

Referenced by find_peer().

{
   struct sip_peer *search = obj, *match = arg;
   int *which_objects = data;

   /* Usernames in SIP uri's are case sensitive. Domains are not */
   if (strcmp(search->name, match->name)) {
      return 0;
   }

   switch (*which_objects) {
   case FINDUSERS:
      if (!(search->type & SIP_TYPE_USER)) {
         return 0;
      }
      break;
   case FINDPEERS:
      if (!(search->type & SIP_TYPE_PEER)) {
         return 0;
      }
      break;
   case FINDALLDEVICES:
      break;
   }

   return CMP_MATCH | CMP_STOP;
}

static int find_by_notify_uri_helper	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 1622 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and ast_cc_agent::private_data.

Referenced by find_sip_cc_agent_by_notify_uri().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   const char *uri = arg;

   return !strcmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static int find_by_subscribe_uri_helper	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 1637 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and ast_cc_agent::private_data.

Referenced by find_sip_cc_agent_by_subscribe_uri().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   const char *uri = arg;

   return !strcmp(agent_pvt->subscribe_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static struct sip_pvt * find_call	(	struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const int	intended_method
	)			[static, read]

find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_incoming(), sipsock_read

Definition at line 7334 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_next, ao2_t_callback, ao2_t_find, args, ast_debug, ast_strdupa, ast_strlen_zero(), match_req_args::authentication_present, match_req_args::callid, dialog_find_multiple(), match_req_args::fromtag, get_header(), gettag(), match_req_to_dialog(), match_req_args::method, OBJ_MULTIPLE, OBJ_POINTER, match_req_args::ruri, match_req_args::seqno, sip_alloc(), sip_cfg, sip_methods, sip_pvt_lock, SIP_REQ_LOOP_DETECTED, SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, cfsip_methods::text, match_req_args::totag, transmit_response_using_temp(), match_req_args::viabranch, and match_req_args::viasentby.

Referenced by handle_request_do().

{
   struct sip_pvt *p = NULL;
   char totag[128];
   char fromtag[128];
   const char *callid = get_header(req, "Call-ID");
   const char *from = get_header(req, "From");
   const char *to = get_header(req, "To");
   const char *cseq = get_header(req, "Cseq");
   struct sip_pvt *sip_pvt_ptr;
   unsigned int seqno;
   /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
   /* get_header always returns non-NULL so we must use ast_strlen_zero() */
   if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
         ast_strlen_zero(from) || ast_strlen_zero(cseq) ||
         (sscanf(cseq, "%30u", &seqno) != 1)) {

      /* RFC 3261 section 24.4.1.   Send a 400 Bad Request if the request is malformed. */
      if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
         transmit_response_using_temp(callid, addr, 1, intended_method,
                       req, "400 Bad Request");
      }
      return NULL;   /* Invalid packet */
   }

   if (sip_cfg.pedanticsipchecking) {
      /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
         we need more to identify a branch - so we have to check branch, from
         and to tags to identify a call leg.
         For Asterisk to behave correctly, you need to turn on pedanticsipchecking
         in sip.conf
         */
      if (gettag(req, "To", totag, sizeof(totag)))
         req->has_to_tag = 1; /* Used in handle_request/response */
      gettag(req, "From", fromtag, sizeof(fromtag));

      ast_debug(5, "= Looking for  Call ID: %s (Checking %s) --From tag %s --To-tag %s  \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);

      /* All messages must always have From: tag */
      if (ast_strlen_zero(fromtag)) {
         ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
      /* reject requests that must always have a To: tag */
      if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
         if (req->method != SIP_ACK) {
            transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
         }
         ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
   }

   if (!sip_cfg.pedanticsipchecking) {
      struct sip_pvt tmp_dialog = {
         .callid = callid,
      };
      sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find in dialogs");
      if (sip_pvt_ptr) {  /* well, if we don't find it-- what IS in there? */
         /* Found the call */
         sip_pvt_lock(sip_pvt_ptr);
         return sip_pvt_ptr;
      }
   } else { /* in pedantic mode! -- do the fancy search */
      struct sip_pvt tmp_dialog = {
         .callid = callid,
      };
      struct match_req_args args = { 0, };
      int found;
      struct ao2_iterator *iterator = ao2_t_callback(dialogs,
         OBJ_POINTER | OBJ_MULTIPLE,
         dialog_find_multiple,
         &tmp_dialog,
         "pedantic ao2_find in dialogs");

      args.method = req->method;
      args.callid = NULL; /* we already matched this. */
      args.totag = totag;
      args.fromtag = fromtag;
      args.seqno = seqno;

      /* If this is a Request, set the Via and Authorization header arguments */
      if (req->method != SIP_RESPONSE) {
         args.ruri = REQ_OFFSET_TO_STR(req, rlPart2);
         get_viabranch(ast_strdupa(get_header(req, "Via")), (char **) &args.viasentby, (char **) &args.viabranch);
         if (!ast_strlen_zero(get_header(req, "Authorization")) ||
            !ast_strlen_zero(get_header(req, "Proxy-Authorization"))) {
            args.authentication_present = 1;
         }
      }

      /* Iterate a list of dialogs already matched by Call-id */
      while (iterator && (sip_pvt_ptr = ao2_iterator_next(iterator))) {
         found = match_req_to_dialog(sip_pvt_ptr, &args);

         switch (found) {
         case SIP_REQ_MATCH:
            sip_pvt_lock(sip_pvt_ptr);
            ao2_iterator_destroy(iterator);
            return sip_pvt_ptr; /* return pvt with ref */
         case SIP_REQ_LOOP_DETECTED:
            /* This is likely a forked Request that somehow resulted in us receiving multiple parts of the fork.
            * RFC 3261 section 8.2.2.2, Indicate that we want to merge requests by sending a 482 response. */
            transmit_response_using_temp(callid, addr, 1, intended_method, req, "482 (Loop Detected)");
            dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search.");
            ao2_iterator_destroy(iterator);
            return NULL;
         case SIP_REQ_NOT_MATCH:
         default:
            dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search");
         }
      }
      if (iterator) {
         ao2_iterator_destroy(iterator);
      }
   } /* end of pedantic mode Request/Reponse to Dialog matching */

   /* See if the method is capable of creating a dialog */
   if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
      if (intended_method == SIP_REFER) {
         /* We do support REFER, but not outside of a dialog yet */
         transmit_response_using_temp(callid, addr, 1, intended_method, req, "603 Declined (no dialog)");
      } else {
         /* Ok, time to create a new SIP dialog object, a pvt */
         if ((p = sip_alloc(callid, addr, 1, intended_method, req)))  {
            /* Ok, we've created a dialog, let's go and process it */
            sip_pvt_lock(p);
         } else {
            /* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
               getting a dialog from sip_alloc.
   
               Without a dialog we can't retransmit and handle ACKs and all that, but at least
               send an error message.
   
               Sorry, we apologize for the inconvienience
            */
            transmit_response_using_temp(callid, addr, 1, intended_method, req, "500 Server internal error");
            ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
         }
      }
      return p; /* can be NULL */
   } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
      /* A method we do not support, let's take it on the volley */
      transmit_response_using_temp(callid, addr, 1, intended_method, req, "501 Method Not Implemented");
      ast_debug(2, "Got a request with unsupported SIP method.\n");
   } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
      /* This is a request outside of a dialog that we don't know about */
      transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
      ast_debug(2, "That's odd...  Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>");
   }
   /* We do not respond to responses for dialogs that we don't know about, we just drop
      the session quickly */
   if (intended_method == SIP_RESPONSE)
      ast_debug(2, "That's odd...  Got a response on a call we don't know about. Callid %s\n", callid ? callid : "<unknown>");

   return NULL;
}

static int find_calling_channel	(	void *	obj,
		void *	arg,
		void *	data,
		int	flags
	)			[static]

Find the channel that is causing the RINGING update.

Definition at line 11536 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, CMP_MATCH, CMP_STOP, ast_channel::context, ast_channel::exten, ast_channel::macroexten, ast_channel::pbx, and sip_cfg.

Referenced by state_notify_build_xml().

{
   struct ast_channel *c = obj;
   struct sip_pvt *p = data;
   int res;

   ast_channel_lock(c);

   res = (c->pbx &&
         (!strcasecmp(c->macroexten, p->exten) || !strcasecmp(c->exten, p->exten)) &&
         (sip_cfg.notifycid == IGNORE_CONTEXT || !strcasecmp(c->context, p->context)));

   ast_channel_unlock(c);

   return res ? CMP_MATCH | CMP_STOP : 0;
}

const char* find_closing_quote	(	const char *	start,
		const char *	lim
	)

Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.

Definition at line 4098 of file chan_sip.c.

Referenced by parse_moved_contact().

{
   char last_char = '\0';
   const char *s;
   for (s = start; *s && s != lim; last_char = *s++) {
      if (*s == '"' && last_char != '\\')
         break;
   }
   return s;
}

static struct sip_peer * find_peer	(	const char *	peer,
		struct ast_sockaddr *	addr,
		int	realtime,
		int	which_objects,
		int	devstate_only,
		int	transport
	)			[static, read]

Locate device by name or ip address.

Parameters:

	peer,sin,realtime,devstate_only,transport
	which_objects	Define which objects should be matched when doing a lookup by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES. Note that this option is not used at all when doing a lookup by IP.

This is used on find matching device on name or ip/port. If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.

Note:

Avoid using this function in new functions if there is a way to avoid it, since it might cause a database lookup.

Definition at line 4594 of file chan_sip.c.

References ao2_t_callback_data, ao2_t_find, ast_copy_string(), ast_set_flag, ast_sockaddr_copy(), find_by_name(), OBJ_POINTER, realtime_peer(), and unref_peer().

{
   struct sip_peer *p = NULL;
   struct sip_peer tmp_peer;

   if (peer) {
      ast_copy_string(tmp_peer.name, peer, sizeof(tmp_peer.name));
      p = ao2_t_callback_data(peers, OBJ_POINTER, find_by_name, &tmp_peer, &which_objects, "ao2_find in peers table");
   } else if (addr) { /* search by addr? */
      ast_sockaddr_copy(&tmp_peer.addr, addr);
      tmp_peer.flags[0].flags = 0;
      tmp_peer.transports = transport;
      p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table"); /* WAS:  p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
      if (!p) {
         ast_set_flag(&tmp_peer.flags[0], SIP_INSECURE_PORT);
         p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table 2"); /* WAS:  p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
         if (p) {
            return p;
         }
      }
   }

   if (!p && (realtime || devstate_only)) {
      p = realtime_peer(peer, addr, devstate_only);
      if (p) {
         switch (which_objects) {
         case FINDUSERS:
            if (!(p->type & SIP_TYPE_USER)) {
               unref_peer(p, "Wrong type of realtime SIP endpoint");
               return NULL;
            }
            break;
         case FINDPEERS:
            if (!(p->type & SIP_TYPE_PEER)) {
               unref_peer(p, "Wrong type of realtime SIP endpoint");
               return NULL;
            }
            break;
         case FINDALLDEVICES:
            break;
         }
      }
   }

   return p;
}

static struct sip_auth * find_realm_authentication	(	struct sip_auth *	authlist,
		const char *	realm
	)			[static, read]

Find authentication for a specific realm.

Definition at line 25419 of file chan_sip.c.

Referenced by build_reply_digest().

{
   struct sip_auth *a;

   for (a = authlist; a; a = a->next) {
      if (!strcasecmp(a->realm, realm))
         break;
   }

   return a;
}

static int find_sdp

(

struct sip_request *

req

)

[static]

Determine whether a SIP message contains an SDP in its body.

Parameters:

req

the SIP request to process

Returns:

1 if SDP found, 0 if not found

Also updates req->sdp_start and req->sdp_count to indicate where the SDP lives in the message body.

Definition at line 7838 of file chan_sip.c.

References ast_log(), ast_strdupa, ast_strlen_zero(), FALSE, get_header(), LOG_WARNING, strcasestr(), and TRUE.

Referenced by handle_incoming(), handle_request_invite(), handle_response(), and handle_response_invite().

{
   const char *content_type;
   const char *content_length;
   const char *search;
   char *boundary;
   unsigned int x;
   int boundaryisquoted = FALSE;
   int found_application_sdp = FALSE;
   int found_end_of_headers = FALSE;

   content_length = get_header(req, "Content-Length");

   if (!ast_strlen_zero(content_length)) {
      if (sscanf(content_length, "%30u", &x) != 1) {
         ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length);
         return 0;
      }

      /* Content-Length of zero means there can't possibly be an
         SDP here, even if the Content-Type says there is */
      if (x == 0)
         return 0;
   }

   content_type = get_header(req, "Content-Type");

   /* if the body contains only SDP, this is easy */
   if (!strncasecmp(content_type, "application/sdp", 15)) {
      req->sdp_start = 0;
      req->sdp_count = req->lines;
      return req->lines ? 1 : 0;
   }

   /* if it's not multipart/mixed, there cannot be an SDP */
   if (strncasecmp(content_type, "multipart/mixed", 15))
      return 0;

   /* if there is no boundary marker, it's invalid */
   if ((search = strcasestr(content_type, ";boundary=")))
      search += 10;
   else if ((search = strcasestr(content_type, "; boundary=")))
      search += 11;
   else
      return 0;

   if (ast_strlen_zero(search))
      return 0;

   /* If the boundary is quoted with ", remove quote */
   if (*search == '\"')  {
      search++;
      boundaryisquoted = TRUE;
   }

   /* make a duplicate of the string, with two extra characters
      at the beginning */
   boundary = ast_strdupa(search - 2);
   boundary[0] = boundary[1] = '-';
   /* Remove final quote */
   if (boundaryisquoted)
      boundary[strlen(boundary) - 1] = '\0';

   /* search for the boundary marker, the empty line delimiting headers from
      sdp part and the end boundry if it exists */

   for (x = 0; x < (req->lines); x++) {
      const char *line = REQ_OFFSET_TO_STR(req, line[x]);
      if (!strncasecmp(line, boundary, strlen(boundary))){
         if (found_application_sdp && found_end_of_headers) {
            req->sdp_count = (x - 1) - req->sdp_start;
            return 1;
         }
         found_application_sdp = FALSE;
      }
      if (!strcasecmp(line, "Content-Type: application/sdp"))
         found_application_sdp = TRUE;
      
      if (ast_strlen_zero(line)) {
         if (found_application_sdp && !found_end_of_headers){
            req->sdp_start = x;
            found_end_of_headers = TRUE;
         }
      }
   }
   if (found_application_sdp && found_end_of_headers) {
      req->sdp_count = x - req->sdp_start;
      return TRUE;
   }
   return FALSE;
}

static struct ast_cc_agent* find_sip_cc_agent_by_notify_uri

(

const char *const

uri

)

[static, read]

Definition at line 1631 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_notify_uri_helper().

Referenced by get_destination().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_notify_uri_helper, (char *)uri, "SIP");
   return agent;
}

static struct ast_cc_agent* find_sip_cc_agent_by_original_callid

(

struct sip_pvt *

pvt

)

[static, read]

Definition at line 1661 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_callid_helper().

Referenced by add_cc_call_info_to_response().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_callid_helper, pvt, "SIP");
   return agent;
}

static struct ast_cc_agent* find_sip_cc_agent_by_subscribe_uri

(

const char *const

uri

)

[static, read]

Definition at line 1646 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_subscribe_uri_helper().

Referenced by handle_cc_subscribe().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_subscribe_uri_helper, (char *)uri, "SIP");
   return agent;
}

static int find_sip_method

(

const char *

msg

)

[static]

find_sip_method: Find SIP method from header

Definition at line 2912 of file chan_sip.c.

References ARRAY_LEN, ast_strlen_zero(), cfsip_methods::id, method_match(), and sip_methods.

Referenced by __sip_pretend_ack(), handle_request_do(), handle_response(), mark_parsed_methods(), and sip_hangup().

{
   int i, res = 0;
   
   if (ast_strlen_zero(msg)) {
      return 0;
   }
   for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
      if (method_match(i, msg)) {
         res = sip_methods[i].id;
      }
   }
   return res;
}

static int find_sip_monitor_instance_by_subscription_pvt	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 1843 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by handle_cc_notify(), and handle_response().

{
   struct sip_monitor_instance *monitor_instance = obj;
   return monitor_instance->subscription_pvt == arg ? CMP_MATCH | CMP_STOP : 0;
}

static int find_sip_monitor_instance_by_suspension_entry	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 1849 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by cc_handle_publish_error().

{
   struct sip_monitor_instance *monitor_instance = obj;
   return monitor_instance->suspension_entry == arg ? CMP_MATCH | CMP_STOP : 0;
}

static struct epa_static_data* find_static_data

(

const char *const

event_package

)

[static, read]

Definition at line 886 of file chan_sip.c.

References AST_LIST_LOCK, AST_LIST_TRAVERSE, and AST_LIST_UNLOCK.

Referenced by create_epa_entry().

{
   const struct epa_backend *backend = NULL;

   AST_LIST_LOCK(&epa_static_data_list);
   AST_LIST_TRAVERSE(&epa_static_data_list, backend, next) {
      if (!strcmp(backend->static_data->name, event_package)) {
         break;
      }
   }
   AST_LIST_UNLOCK(&epa_static_data_list);
   return backend ? backend->static_data : NULL;
}

static struct cfsubscription_types * find_subscription_type

(

enum subscriptiontype

subtype

)

[static, read]

Find subscription type in array.

Definition at line 17034 of file chan_sip.c.

References ARRAY_LEN, subscription_types, and type.

Referenced by transmit_state_notify().

{
   int i;

   for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
      if (subscription_types[i].type == subtype) {
         return &subscription_types[i];
      }
   }
   return &subscription_types[0];
}

static void free_old_route

(

struct sip_route *

route

)

[static]

Remove route from route list.

Definition at line 13100 of file chan_sip.c.

References ast_free.

Referenced by __sip_destroy(), and build_route().

{
   struct sip_route *next;

   while (route) {
      next = route->next;
      ast_free(route);
      route = next;
   }
}

static int func_check_sipdomain	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

Dial plan function to check if domain is local.

Definition at line 18154 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strlen_zero(), check_sip_domain(), and LOG_WARNING.

{
   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
      return -1;
   }
   if (check_sip_domain(data, NULL, 0))
      ast_copy_string(buf, data, len);
   else
      buf[0] = '\0';
   return 0;
}

static int func_header_read	(	struct ast_channel *	chan,
		const char *	function,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

Read SIP header (dialplan function).

Definition at line 18095 of file chan_sip.c.

References __get_header(), args, AST_APP_ARG, ast_channel_lock, ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_log(), AST_STANDARD_APP_ARGS, ast_strlen_zero(), LOG_WARNING, ast_channel::tech, and ast_channel::tech_pvt.

{
   struct sip_pvt *p;
   const char *content = NULL;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(header);
      AST_APP_ARG(number);
   );
   int i, number, start = 0;

   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "This function requires a header name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   AST_STANDARD_APP_ARGS(args, data);
   if (!args.number) {
      number = 1;
   } else {
      sscanf(args.number, "%30d", &number);
      if (number < 1)
         number = 1;
   }

   p = chan->tech_pvt;

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   for (i = 0; i < number; i++)
      content = __get_header(&p->initreq, args.header, &start);

   if (ast_strlen_zero(content)) {
      ast_channel_unlock(chan);
      return -1;
   }

   ast_copy_string(buf, content, len);
   ast_channel_unlock(chan);

   return 0;
}

static int function_sipchaninfo_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

${SIPCHANINFO()} Dialplan function - reads sip channel data

Definition at line 18272 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, ast_copy_string(), ast_log(), ast_sockaddr_stringify_addr(), LOG_WARNING, ast_channel::tech, and ast_channel::tech_pvt.

{
   struct sip_pvt *p;
   static int deprecated = 0;

   *buf = 0;
   
   if (!data) {
      ast_log(LOG_WARNING, "This function requires a parameter name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(chan->tech)) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   if (deprecated++ % 20 == 0) {
      /* Deprecated in 1.6.1 */
      ast_log(LOG_WARNING, "SIPCHANINFO() is deprecated.  Please transition to using CHANNEL().\n");
   }

   p = chan->tech_pvt;

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   if (!strcasecmp(data, "peerip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->sa), len);
   } else  if (!strcasecmp(data, "recvip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->recv), len);
   } else  if (!strcasecmp(data, "from")) {
      ast_copy_string(buf, p->from, len);
   } else  if (!strcasecmp(data, "uri")) {
      ast_copy_string(buf, p->uri, len);
   } else  if (!strcasecmp(data, "useragent")) {
      ast_copy_string(buf, p->useragent, len);
   } else  if (!strcasecmp(data, "peername")) {
      ast_copy_string(buf, p->peername, len);
   } else if (!strcasecmp(data, "t38passthrough")) {
      if (p->t38.state == T38_DISABLED) {
         ast_copy_string(buf, "0", len);
      } else { /* T38 is offered or enabled in this call */
         ast_copy_string(buf, "1", len);
      }
   } else {
      ast_channel_unlock(chan);
      return -1;
   }
   ast_channel_unlock(chan);

   return 0;
}

static int function_sippeer	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)			[static]

${SIPPEER()} Dialplan function - reads peer data

Todo:

Will be deprecated after 1.4

Definition at line 18173 of file chan_sip.c.

References ast_codec_pref_index(), ast_copy_string(), ast_getformatname(), ast_getformatname_multiple(), ast_log(), ast_print_group(), ast_sockaddr_port, ast_sockaddr_stringify_addr(), ast_str_alloca, ast_test_flag, chanvar, FALSE, find_peer(), LOG_WARNING, ast_variable::name, ast_variable::next, peer_mailboxes_to_str(), peer_status(), strsep(), TRUE, unref_peer(), and ast_variable::value.

{
   struct sip_peer *peer;
   char *colname;

   if ((colname = strchr(data, ':'))) {   /*! \todo Will be deprecated after 1.4 */
      static int deprecation_warning = 0;
      *colname++ = '\0';
      if (deprecation_warning++ % 10 == 0)
         ast_log(LOG_WARNING, "SIPPEER(): usage of ':' to separate arguments is deprecated.  Please use ',' instead.\n");
   } else if ((colname = strchr(data, ',')))
      *colname++ = '\0';
   else
      colname = "ip";

   if (!(peer = find_peer(data, NULL, TRUE, FINDPEERS, FALSE, 0)))
      return -1;

   if (!strcasecmp(colname, "ip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&peer->addr), len);
   } else  if (!strcasecmp(colname, "port")) {
      snprintf(buf, len, "%d", ast_sockaddr_port(&peer->addr));
   } else  if (!strcasecmp(colname, "status")) {
      peer_status(peer, buf, len);
   } else  if (!strcasecmp(colname, "language")) {
      ast_copy_string(buf, peer->language, len);
   } else  if (!strcasecmp(colname, "regexten")) {
      ast_copy_string(buf, peer->regexten, len);
   } else  if (!strcasecmp(colname, "limit")) {
      snprintf(buf, len, "%d", peer->call_limit);
   } else  if (!strcasecmp(colname, "busylevel")) {
      snprintf(buf, len, "%d", peer->busy_level);
   } else  if (!strcasecmp(colname, "curcalls")) {
      snprintf(buf, len, "%d", peer->inUse);
   } else if (!strcasecmp(colname, "maxforwards")) {
      snprintf(buf, len, "%d", peer->maxforwards);
   } else  if (!strcasecmp(colname, "accountcode")) {
      ast_copy_string(buf, peer->accountcode, len);
   } else  if (!strcasecmp(colname, "callgroup")) {
      ast_print_group(buf, len, peer->callgroup);
   } else  if (!strcasecmp(colname, "pickupgroup")) {
      ast_print_group(buf, len, peer->pickupgroup);
   } else  if (!strcasecmp(colname, "useragent")) {
      ast_copy_string(buf, peer->useragent, len);
   } else  if (!strcasecmp(colname, "mailbox")) {
      struct ast_str *mailbox_str = ast_str_alloca(512);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_copy_string(buf, mailbox_str->str, len);
   } else  if (!strcasecmp(colname, "context")) {
      ast_copy_string(buf, peer->context, len);
   } else  if (!strcasecmp(colname, "expire")) {
      snprintf(buf, len, "%d", peer->expire);
   } else  if (!strcasecmp(colname, "dynamic")) {
      ast_copy_string(buf, peer->host_dynamic ? "yes" : "no", len);
   } else  if (!strcasecmp(colname, "callerid_name")) {
      ast_copy_string(buf, peer->cid_name, len);
   } else  if (!strcasecmp(colname, "callerid_num")) {
      ast_copy_string(buf, peer->cid_num, len);
   } else  if (!strcasecmp(colname, "codecs")) {
      ast_getformatname_multiple(buf, len -1, peer->capability);
   } else if (!strcasecmp(colname, "encryption")) {
      snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
   } else  if (!strncasecmp(colname, "chanvar[", 8)) {
      char *chanvar=colname + 8;
      struct ast_variable *v;
   
      chanvar = strsep(&chanvar, "]");
      for (v = peer->chanvars ; v ; v = v->next) {
         if (!strcasecmp(v->name, chanvar)) {
            ast_copy_string(buf, v->value, len);
         }
      }
   } else  if (!strncasecmp(colname, "codec[", 6)) {
      char *codecnum;
      format_t codec = 0;
      
      codecnum = colname + 6; /* move past the '[' */
      codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
      if((codec = ast_codec_pref_index(&peer->prefs, atoi(codecnum)))) {
         ast_copy_string(buf, ast_getformatname(codec), len);
      } else {
         buf[0] = '\0';
      }
   } else {
      buf[0] = '\0';
   }

   unref_peer(peer, "unref_peer from function_sippeer, just before return");

   return 0;
}

static char * generate_random_string	(	char *	buf,
		size_t	size
	)			[static]

Generate 32 byte random string for callid's etc.

Definition at line 6957 of file chan_sip.c.

References ast_random().

Referenced by build_callid_pvt(), build_callid_registry(), calendar_query_exec(), construct_pidf_body(), do_notify(), and generate_uri().

{
   long val[4];
   int x;

   for (x=0; x<4; x++)
      val[x] = ast_random();
   snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);

   return buf;
}

static char* generate_uri	(	struct sip_pvt *	pvt,
		char *	buf,
		size_t	size
	)			[static]

Definition at line 6969 of file chan_sip.c.

References ast_copy_string(), ast_sockaddr_stringify(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_set(), and generate_random_string().

Referenced by add_cc_call_info_to_response(), and transmit_cc_notify().

{
   struct ast_str *uri = ast_str_alloca(size);
   ast_str_set(&uri, 0, "%s", pvt->socket.type == SIP_TRANSPORT_TLS ? "sips:" : "sip:");
   /* Here would be a great place to generate a UUID, but for now we'll
    * use the handy random string generation function we already have
    */
   ast_str_append(&uri, 0, "%s", generate_random_string(buf, size));
   ast_str_append(&uri, 0, "@%s", ast_sockaddr_stringify(&pvt->ourip));
   ast_copy_string(buf, ast_str_buffer(uri), size);
   return buf;
}

int get_address_family_filter

(

const struct ast_sockaddr *

addr

)

[static]

Helper for dns resolution to filter by address family.

Note:

return 0 if addr is [::] else it returns addr's family.

Definition at line 23938 of file chan_sip.c.

References ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), and ast_sockaddr::ss.

Referenced by __sip_subscribe_mwi_do(), ast_sockaddr_resolve_first(), build_peer(), proxy_update(), realtime_peer(), and transmit_register().

{
   if (ast_sockaddr_is_ipv6(addr) && ast_sockaddr_is_any(addr)) {
      return 0;
   }

   return addr->ss.ss_family;
}

static int get_also_info	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)			[static]

Call transfer support (old way, deprecated by the IETF).

Note:

does not account for SIPS: uri requirements, nor check transport

Definition at line 14644 of file chan_sip.c.

References ast_canmatch_extension(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_log(), ast_string_field_set, ast_strlen_zero(), ast_verbose, context, get_header(), LOG_WARNING, pbx_builtin_getvar_helper(), S_OR, sip_cfg, sip_debug_test_pvt(), SIP_PEDANTIC_DECODE, and sip_refer_allocate().

Referenced by handle_request_bye().

{
   char tmp[256] = "", *c, *a;
   struct sip_request *req = oreq ? oreq : &p->initreq;
   struct sip_refer *referdata = NULL;
   const char *transfer_context = NULL;
   
   if (!p->refer && !sip_refer_allocate(p))
      return -1;

   referdata = p->refer;

   ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
   c = get_in_brackets(tmp);

   if (parse_uri(c, "sip:,sips:", &c, NULL, &a, NULL)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header in Also: transfer (%s)?\n", c);
      return -1;
   }
   
   SIP_PEDANTIC_DECODE(c);
   SIP_PEDANTIC_DECODE(a);

   if (!ast_strlen_zero(a)) {
      ast_copy_string(referdata->refer_to_domain, a, sizeof(referdata->refer_to_domain));
   }

   if (sip_debug_test_pvt(p))
      ast_verbose("Looking for %s in %s\n", c, p->context);

   if (p->owner)  /* Mimic behaviour in res_features.c */
      transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");

   /* By default, use the context in the channel sending the REFER */
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(p->owner->macrocontext,
               S_OR(p->context, sip_cfg.default_context));
   }
   if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
      /* This is a blind transfer */
      ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
      ast_copy_string(referdata->refer_to, c, sizeof(referdata->refer_to));
      ast_copy_string(referdata->referred_by, "", sizeof(referdata->referred_by));
      ast_copy_string(referdata->refer_contact, "", sizeof(referdata->refer_contact));
      referdata->refer_call = dialog_unref(referdata->refer_call, "unreffing referdata->refer_call");
      /* Set new context */
      ast_string_field_set(p, context, transfer_context);
      return 0;
   } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
      return 1;
   }

   return -1;
}

static char* get_body	(	struct sip_request *	req,
		char *	name,
		char	delimiter
	)			[static]

Get a specific line from the message body.

Definition at line 6724 of file chan_sip.c.

References get_body_by_line(), and len().

Referenced by handle_cc_notify(), handle_request_info(), and handle_request_notify().

{
   int x;
   int len = strlen(name);
   char *r;

   for (x = 0; x < req->lines; x++) {
      r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[x]), name, len, delimiter);
      if (r[0] != '\0')
         return r;
   }

   return "";
}

static char* get_body_by_line	(	const char *	line,
		const char *	name,
		int	nameLen,
		char	delimiter
	)			[static]

Reads one line of SIP message body.

Definition at line 6669 of file chan_sip.c.

References ast_skip_blanks().

Referenced by get_body(), and get_sdp_iterate().

{
   if (!strncasecmp(line, name, nameLen) && line[nameLen] == delimiter)
      return ast_skip_blanks(line + nameLen + 1);

   return "";
}

static int get_cached_mwi	(	struct sip_peer *	peer,
		int *	new,
		int *	old
	)			[static]

Get cached MWI info.

Return values:

	0	At least one message is waiting
	1	no messages waiting

Definition at line 24081 of file chan_sip.c.

References ast_event_destroy(), ast_event_get_cached(), ast_event_get_ie_uint(), AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, AST_LIST_TRAVERSE, mailbox, and S_OR.

Referenced by sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      struct ast_event *event;
      event = ast_event_get_cached(AST_EVENT_MWI,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
      if (!event)
         continue;
      *new += ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
      *old += ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
      ast_event_destroy(event);
   }

   return (*new || *old) ? 0 : 1;
}

static void get_crypto_attrib	(	struct sip_srtp *	srtp,
		const char **	a_crypto
	)			[static]

Definition at line 10382 of file chan_sip.c.

References ast_log(), and LOG_WARNING.

Referenced by add_sdp().

{
   /* Set encryption properties */
   if (srtp) {
      if (!srtp->crypto) {
         srtp->crypto = sdp_crypto_setup();
      }
      if (srtp->crypto && (sdp_crypto_offer(srtp->crypto) >= 0)) {
         *a_crypto = sdp_crypto_attrib(srtp->crypto);
      }

      if (!*a_crypto) {
         ast_log(LOG_WARNING, "No SRTP key management enabled\n");
      }
   }
}

static enum sip_get_dest_result get_destination	(	struct sip_pvt *	p,
		struct sip_request *	oreq,
		int *	cc_recall_core_id
	)			[static]

Find out who the call is for. We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.

Returns:

0 on success (found a matching extension), non-zero on failure

Note:

If the incoming uri is a SIPS: uri, we are required to carry this across the dialplan, so that the outbound call also is a sips: call or encrypted IAX2 call. If that's not available, the call should FAIL.

Definition at line 14230 of file chan_sip.c.

References ao2_ref, ast_canmatch_extension(), ast_cc_agent_recalling(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_get_hint(), AST_LIST_EMPTY, ast_log(), AST_MAX_EXTENSION, ast_pickup_ext(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_verbose, check_sip_domain(), context, ast_cc_agent::core_id, ast_cc_agent::device_name, dummy(), exten, find_sip_cc_agent_by_notify_uri(), get_header(), LOG_WARNING, ast_cc_agent::private_data, S_OR, sip_cfg, sip_debug_test_pvt(), sip_methods, SIP_PEDANTIC_DECODE, and cfsip_methods::text.

Referenced by handle_request_invite(), handle_request_options(), and handle_request_subscribe().

{
   char tmp[256] = "", *uri, *domain, *dummy = NULL;
   char tmpf[256] = "", *from = NULL;
   struct sip_request *req;
   char *decoded_uri;

   req = oreq;
   if (!req) {
      req = &p->initreq;
   }

   /* Find the request URI */
   if (req->rlPart2)
      ast_copy_string(tmp, REQ_OFFSET_TO_STR(req, rlPart2), sizeof(tmp));
   
   uri = ast_strdupa(get_in_brackets(tmp));

   if (parse_uri(uri, "sip:,sips:", &uri, &dummy, &domain, NULL)) {
      ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", uri);
      return SIP_GET_DEST_INVALID_URI;
   }

   SIP_PEDANTIC_DECODE(domain);
   SIP_PEDANTIC_DECODE(uri);

   ast_string_field_set(p, domain, domain);

   /* Now find the From: caller ID and name */
   /* XXX Why is this done in get_destination? Isn't it already done?
      Needs to be checked
        */
   ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
   if (!ast_strlen_zero(tmpf)) {
      from = get_in_brackets(tmpf);
      if (parse_uri(from, "sip:,sips:", &from, NULL, &domain, NULL)) {
         ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", from);
         return SIP_GET_DEST_INVALID_URI;
      }

      SIP_PEDANTIC_DECODE(from);
      SIP_PEDANTIC_DECODE(domain);

      ast_string_field_set(p, fromdomain, domain);
   }

   if (!AST_LIST_EMPTY(&domain_list)) {
      char domain_context[AST_MAX_EXTENSION];

      domain_context[0] = '\0';
      if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
         if (!sip_cfg.allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
            ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
            return SIP_GET_DEST_REFUSED;
         }
      }
      /* If we don't have a peer (i.e. we're a guest call),
       * overwrite the original context */
      if (!ast_test_flag(&p->flags[1], SIP_PAGE2_HAVEPEERCONTEXT) && !ast_strlen_zero(domain_context)) {
         ast_string_field_set(p, context, domain_context);
      }
   }

   /* If the request coming in is a subscription and subscribecontext has been specified use it */
   if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext)) {
      ast_string_field_set(p, context, p->subscribecontext);
   }

   if (sip_debug_test_pvt(p)) {
      ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain);
   }

   /* If this is a subscription we actually just need to see if a hint exists for the extension */
   if (req->method == SIP_SUBSCRIBE) {
      char hint[AST_MAX_EXTENSION];
      return (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten) ?
         SIP_GET_DEST_EXTEN_FOUND :
         SIP_GET_DEST_EXTEN_NOT_FOUND);
   } else {
      struct ast_cc_agent *agent;
      decoded_uri = ast_strdupa(uri);
      ast_uri_decode(decoded_uri);
      /* Check the dialplan for the username part of the request URI,
         the domain will be stored in the SIPDOMAIN variable
         Since extensions.conf can have unescaped characters, try matching a decoded
         uri in addition to the non-decoded uri
         Return 0 if we have a matching extension */
      if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) ||
         ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) ||
          !strcmp(decoded_uri, ast_pickup_ext())) {
         if (!oreq) {
            ast_string_field_set(p, exten, decoded_uri);
         }
         return SIP_GET_DEST_EXTEN_FOUND;
      } else if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
         struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
         /* This is a CC recall. We can set p's extension to the exten from
          * the original INVITE
          */
         ast_string_field_set(p, exten, agent_pvt->original_exten);
         /* And we need to let the CC core know that the caller is attempting
          * his recall
          */
         ast_cc_agent_recalling(agent->core_id, "SIP caller %s is attempting recall",
               agent->device_name);
         if (cc_recall_core_id) {
            *cc_recall_core_id = agent->core_id;
         }
         ao2_ref(agent, -1);
         return SIP_GET_DEST_EXTEN_FOUND;
      }
   }

   /* Return 1 for pickup extension or overlap dialling support (if we support it) */
   if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) &&
       ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) ||
       !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) {
      return SIP_GET_DEST_PICKUP_EXTEN_FOUND;
   }

   return SIP_GET_DEST_EXTEN_NOT_FOUND;
}

static int get_domain	(	const char *	str,
		char *	domain,
		int	len
	)			[static]

Extract domain from SIP To/From header.

Returns:

-1 on error, 1 if domain string is empty, 0 if domain was properly extracted

Note:

TODO: Such code is all over SIP channel, there is a sense to organize this patern in one function

Definition at line 9889 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strlen_zero(), and LOG_WARNING.

Referenced by get_realm().

{
   char tmpf[256];
   char *a, *from;

   *domain = '\0';
   ast_copy_string(tmpf, str, sizeof(tmpf));
   from = get_in_brackets(tmpf);
   if (!ast_strlen_zero(from)) {
      if (strncasecmp(from, "sip:", 4)) {
         ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", from);
         return -1;
      }
      from += 4;
   } else
      from = NULL;

   if (from) {
      int bracket = 0;

      /* Strip any params or options from user */
      if ((a = strchr(from, ';')))
         *a = '\0';
      /* Strip port from domain if present */
      for (a = from; *a != '\0'; ++a) {
         if (*a == ':' && bracket == 0) {
            *a = '\0';
            break;
         } else if (*a == '[') {
            ++bracket;
         } else if (*a == ']') {
            --bracket;
         }
      }
      if ((a = strchr(from, '@'))) {
         *a = '\0';
         ast_copy_string(domain, a + 1, len);
      } else
         ast_copy_string(domain, from, len);
   }

   return ast_strlen_zero(domain);
}

static struct event_state_compositor* get_esc

(

const char *const

event_package

)

[static, read]

Definition at line 980 of file chan_sip.c.

References ARRAY_LEN, event_state_compositors, and name.

Referenced by create_new_sip_etag(), handle_request_publish(), and publish_expire().

                                                                                {
   int i;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      if (!strcasecmp(event_package, event_state_compositors[i].name)) {
         return &event_state_compositors[i];
      }
   }
   return NULL;
}

static struct sip_esc_entry* get_esc_entry	(	const char *	entity_tag,
		struct event_state_compositor *	esc
	)			[static, read]

Definition at line 990 of file chan_sip.c.

References ao2_find, ast_copy_string(), event_state_compositor::compositor, and OBJ_POINTER.

Referenced by handle_sip_publish_modify(), handle_sip_publish_refresh(), and handle_sip_publish_remove().

                                                                                                        {
   struct sip_esc_entry *entry;
   struct sip_esc_entry finder;

   ast_copy_string(finder.entity_tag, entity_tag, sizeof(finder.entity_tag));

   entry = ao2_find(esc->compositor, &finder, OBJ_POINTER);

   return entry;
}

static const char * get_header	(	const struct sip_request *	req,
		const char *	name
	)			[static]

Get header from SIP request.

Returns:

Always return something, so don't check for NULL because it won't happen :-)

Definition at line 6817 of file chan_sip.c.

References __get_header().

{
   int start = 0;
   return __get_header(req, name, &start);
}

static struct ast_variable * get_insecure_variable_from_config

(

struct ast_config *

config

)

[static, read]

Definition at line 4341 of file chan_sip.c.

References ast_category_browse(), ast_category_root(), ast_test_flag, ast_variable_retrieve(), set_insecure_flags(), and var.

Referenced by realtime_peer().

{
   struct ast_variable *var = NULL;
   struct ast_flags flags = {0};
   char *cat = NULL;
   const char *insecure;
   while ((cat = ast_category_browse(cfg, cat))) {
      insecure = ast_variable_retrieve(cfg, cat, "insecure");
      set_insecure_flags(&flags, insecure, -1);
      if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
         var = ast_category_root(cfg, cat);
         break;
      }
   }
   return var;
}

static int get_ip_and_port_from_sdp	(	struct sip_request *	req,
		const enum media_type	media,
		struct ast_sockaddr *	addr
	)			[static]

Definition at line 7931 of file chan_sip.c.

References ast_log(), ast_sockaddr_resolve_first_af(), ast_strlen_zero(), get_sdp_iterate(), len(), and LOG_WARNING.

Referenced by handle_request_invite().

{
   const char *m;
   const char *c;
   int miterator = req->sdp_start;
   int citerator = req->sdp_start;
   int x = 0;
   int numberofports;
   int len;
   int af;
   char proto[4], host[258] = ""; /*Initialize to empty so we will know if we have any input */

   c = get_sdp_iterate(&citerator, req, "c");
   if (sscanf(c, "IN %3s %256s", proto, host) != 2) {
         ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
         /* Continue since there may be a valid host in a c= line specific to the audio stream */
   }
   /* We only want the m and c lines for audio */
   for (m = get_sdp_iterate(&miterator, req, "m"); !ast_strlen_zero(m); m = get_sdp_iterate(&miterator, req, "m")) {
      if ((media == SDP_AUDIO && ((sscanf(m, "audio %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "audio %30u RTP/AVP %n", &x, &len) == 1 && len > 0))) ||
         (media == SDP_VIDEO && ((sscanf(m, "video %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "video %30u RTP/AVP %n", &x, &len) == 1 && len > 0)))) {
         /* See if there's a c= line for this media stream.
          * XXX There is no guarantee that we'll be grabbing the c= line for this
          * particular media stream here. However, this is the same logic used in process_sdp.
          */
         c = get_sdp_iterate(&citerator, req, "c");
         if (!ast_strlen_zero(c)) {
            sscanf(c, "IN %3s %256s", proto, host);
         }
         break;
      }
   }

   if (!strcmp("IP4", proto)) {
      af = AF_INET;
   } else if (!strcmp("IP6", proto)) {
      af = AF_INET6;
   } else {
      ast_log(LOG_WARNING, "Unknown protocol '%s'.\n", proto);
      return -1;
   }

   if (ast_strlen_zero(host) || x == 0) {
      ast_log(LOG_WARNING, "Failed to read an alternate host or port in SDP. Expect %s problems\n", media == SDP_AUDIO ? "audio" : "video");
      return -1;
   }

   if (ast_sockaddr_resolve_first_af(addr, host, 0, af)) {
      ast_log(LOG_WARNING, "Could not look up IP address of alternate hostname. Expect %s problems\n", media == SDP_AUDIO? "audio" : "video");
      return -1;
   }

   return 0;
}

static int get_msg_text	(	char *	buf,
		int	len,
		struct sip_request *	req,
		int	addnewline
	)			[static]

Get text out of a SIP MESSAGE packet.

Definition at line 15106 of file chan_sip.c.

Referenced by handle_request_info(), handle_request_notify(), and receive_message().

{
   int x;
   int y;

   buf[0] = '\0';
   /*XXX isn't strlen(buf) going to always be 0? */
   y = len - strlen(buf) - 5;
   if (y < 0)
      y = 0;
   for (x = 0; x < req->lines; x++) {
      const char *line = REQ_OFFSET_TO_STR(req, line[x]);
      strncat(buf, line, y); /* safe */
      y -= strlen(line) + 1;
      if (y < 0)
         y = 0;
      if (y != 0 && addnewline)
         strcat(buf, "\n"); /* safe */
   }
   return 0;
}

static const char * get_name_from_variable	(	struct ast_variable *	var,
		const char *	newpeername
	)			[static]

Definition at line 4358 of file chan_sip.c.

References ast_variable::name, ast_variable::next, and ast_variable::value.

Referenced by realtime_peer().

{
   struct ast_variable *tmp;
   for (tmp = var; tmp; tmp = tmp->next) {
      if (!newpeername && !strcasecmp(tmp->name, "name"))
         newpeername = tmp->value;
   }
   return newpeername;
}

static void get_our_media_address	(	struct sip_pvt *	p,
		int	needvideo,
		int	needtext,
		struct ast_sockaddr *	addr,
		struct ast_sockaddr *	vaddr,
		struct ast_sockaddr *	taddr,
		struct ast_sockaddr *	dest,
		struct ast_sockaddr *	vdest,
		struct ast_sockaddr *	tdest
	)			[static]

Set all IP media addresses for this call.

Note:

called from add_sdp()

Definition at line 10289 of file chan_sip.c.

References ast_rtp_instance_get_local_address(), ast_sockaddr_cmp_addr(), ast_sockaddr_copy(), ast_sockaddr_is_any(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, and media_address.

Referenced by add_sdp().

{
   int use_externip = 0;

   /* First, get our address */
   ast_rtp_instance_get_local_address(p->rtp, addr);
   if (p->vrtp) {
      ast_rtp_instance_get_local_address(p->vrtp, vaddr);
   }
   if (p->trtp) {
      ast_rtp_instance_get_local_address(p->trtp, taddr);
   }

   /* If our real IP differs from the local address returned by the RTP engine, use it. */
   /* The premise is that if we are already using that IP to communicate with the client, */
   /* we should be using it for RTP too. */
        use_externip = ast_sockaddr_cmp_addr(&p->ourip, addr);

   /* Now, try to figure out where we want them to send data */
   /* Is this a re-invite to move the media out, then use the original offer from caller  */
   if (!ast_sockaddr_isnull(&p->redirip)) {  /* If we have a redirection IP, use it */
      ast_sockaddr_copy(dest, &p->redirip);
   } else {
      /*
       * Audio Destination IP:
       *
       * 1. Specifically configured media address.
       * 2. Local address as specified by the RTP engine.
       * 3. The local IP as defined by chan_sip.
       *
       * Audio Destination Port:
       *
       * 1. Provided by the RTP engine.
       */
      ast_sockaddr_copy(dest,
              !ast_sockaddr_isnull(&media_address) ? &media_address :
              !ast_sockaddr_is_any(addr) && !use_externip ? addr    :
              &p->ourip);
      ast_sockaddr_set_port(dest, ast_sockaddr_port(addr));
   }

   if (needvideo) {
      /* Determine video destination */
      if (!ast_sockaddr_isnull(&p->vredirip)) {
         ast_sockaddr_copy(vdest, &p->vredirip);
      } else {
         /*
          * Video Destination IP:
          *
          * 1. Specifically configured media address.
          * 2. Local address as specified by the RTP engine.
          * 3. The local IP as defined by chan_sip.
          *
          * Video Destination Port:
          *
          * 1. Provided by the RTP engine.
          */
         ast_sockaddr_copy(vdest,
                 !ast_sockaddr_isnull(&media_address) ? &media_address :
                 !ast_sockaddr_is_any(vaddr) && !use_externip ? vaddr  :
                 &p->ourip);
         ast_sockaddr_set_port(vdest, ast_sockaddr_port(vaddr));
      }
   }

   if (needtext) {
      /* Determine text destination */
      if (!ast_sockaddr_isnull(&p->tredirip)) {
         ast_sockaddr_copy(tdest, &p->tredirip);
      } else {
         /*
          * Text Destination IP:
          *
          * 1. Specifically configured media address.
          * 2. Local address as specified by the RTP engine.
          * 3. The local IP as defined by chan_sip.
          *
          * Text Destination Port:
          *
          * 1. Provided by the RTP engine.
          */
         ast_sockaddr_copy(tdest,
                 !ast_sockaddr_isnull(&media_address) ? &media_address  :
                 !ast_sockaddr_is_any(taddr) && !use_externip ? taddr   :
                 &p->ourip);
         ast_sockaddr_set_port(tdest, ast_sockaddr_port(taddr));
      }
   }
}

static int get_pai	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Parse the parts of the P-Asserted-Identity header on an incoming packet. Returns 1 if a valid header is found and it is different from the current caller id.

Definition at line 13951 of file chan_sip.c.

References ast_copy_string(), ast_is_shrinkable_phonenumber(), AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, ast_set_callerid(), ast_shrink_phone_number(), ast_skip_blanks(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), cid_name, cid_num, and get_header().

Referenced by get_rpid().

{
   char pai[256];
   char privacy[64];
   char *cid_num = "";
   char *cid_name = "";
   int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
   char *start = NULL, *end = NULL, *uri = NULL;

   ast_copy_string(pai, get_header(req, "P-Asserted-Identity"), sizeof(pai));

   if (ast_strlen_zero(pai)) {
      return 0;
   }

   start = pai;
   if (*start == '"') {
      *start++ = '\0';
      end = strchr(start, '"');
      if (!end) {
         return 0;
      }
      *end++ = '\0';
      cid_name = start;
      start = ast_skip_blanks(end);
   }

   if (*start != '<')
      return 0;
   /* At this point, 'start' points to the URI in brackets.
    * We need a copy so that our comparison to the anonymous
    * URI is valid.
    */
   uri = ast_strdupa(start);
   *start++ = '\0';
   end = strchr(start, '@');
   if (!end) {
      return 0;
   }
   *end++ = '\0';
   if (!strncasecmp(uri, "anonymous@anonymous.invalid", 27)) {
      callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
      /*XXX Assume no change in cid_num. Perhaps it should be
       * blanked?
       */
      cid_num = (char *)p->cid_num;
   } else if (!strncasecmp(start, "sip:", 4)) {
      cid_num = start + 4;
      if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num))
         ast_shrink_phone_number(cid_num);
      start = end;

      end = strchr(start, '>');
      if (!end) {
         return 0;
      }
      *end = '\0';
   } else {
      return 0;
   }

   ast_copy_string(privacy, get_header(req, "Privacy"), sizeof(privacy));
   if (!ast_strlen_zero(privacy) && strncmp(privacy, "id", 2)) {
      callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
   }

   /* Only return true if the supplied caller id is different */
   if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres) {
      return 0;
   }

   ast_string_field_set(p, cid_num, cid_num);
   ast_string_field_set(p, cid_name, cid_name);
   p->callingpres = callingpres;

   if (p->owner) {
      ast_set_callerid(p->owner, cid_num, cid_name, NULL);
      p->owner->caller.id.name.presentation = callingpres;
      p->owner->caller.id.number.presentation = callingpres;
   }

   return 1;
}

static int get_rdnis	(	struct sip_pvt *	p,
		struct sip_request *	oreq,
		char **	name,
		char **	number,
		int *	reason
	)			[static]

Get referring dnis.

Definition at line 14136 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strdup, ast_strip_quoted(), ast_strlen_zero(), ast_verbose, exten, get_header(), LOG_WARNING, pbx_builtin_setvar_helper(), sip_debug_test_pvt(), sip_reason_str_to_code(), sip_set_redirstr(), strcasestr(), and strsep().

Referenced by change_redirecting_information().

{
   char tmp[256], *exten, *rexten, *rdomain, *rname = NULL;
   char *params, *reason_param = NULL;
   struct sip_request *req;

   req = oreq ? oreq : &p->initreq;

   ast_copy_string(tmp, get_header(req, "Diversion"), sizeof(tmp));
   if (ast_strlen_zero(tmp))
      return -1;

   if ((params = strchr(tmp, '>'))) {
      params = strchr(params, ';');
   }

   exten = get_in_brackets(tmp);
   if (!strncasecmp(exten, "sip:", 4)) {
      exten += 4;
   } else if (!strncasecmp(exten, "sips:", 5)) {
      exten += 5;
   } else {
      ast_log(LOG_WARNING, "Huh?  Not an RDNIS SIP header (%s)?\n", exten);
      return -1;
   }

   /* Get diversion-reason param if present */
   if (params) {
      *params = '\0';   /* Cut off parameters  */
      params++;
      while (*params == ';' || *params == ' ')
         params++;
      /* Check if we have a reason parameter */
      if ((reason_param = strcasestr(params, "reason="))) {
         char *end;
         reason_param+=7;
         if ((end = strchr(reason_param, ';'))) {
            *end = '\0';
         }
         /* Remove enclosing double-quotes */
         if (*reason_param == '"')
            ast_strip_quoted(reason_param, "\"", "\"");
         if (!ast_strlen_zero(reason_param)) {
            sip_set_redirstr(p, reason_param);
            if (p->owner) {
               pbx_builtin_setvar_helper(p->owner, "__PRIREDIRECTREASON", p->redircause);
               pbx_builtin_setvar_helper(p->owner, "__SIPREDIRECTREASON", reason_param);
            }
         }
      }
   }

   rdomain = exten;
   rexten = strsep(&rdomain, "@");  /* trim anything after @ */
   if (p->owner)
      pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);

   if (sip_debug_test_pvt(p))
      ast_verbose("RDNIS for this call is %s (reason %s)\n", exten, reason ? reason_param : "");

   /*ast_string_field_set(p, rdnis, rexten);*/

   if (*tmp == '\"') {
      char *end_quote;
      rname = tmp + 1;
      end_quote = strchr(rname, '\"');
      *end_quote = '\0';
   }

   if (number) {
      *number = ast_strdup(rexten);
   }

   if (name && rname) {
      *name = ast_strdup(rname);
   }

   if (reason && !ast_strlen_zero(reason_param)) {
      *reason = sip_reason_str_to_code(reason_param);
   }

   return 0;
}

static void get_realm	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)			[static]

Choose realm based on From header and then To header or use globaly configured realm. Realm from From/To header should be listed among served domains in config file: domain=...

Definition at line 9937 of file chan_sip.c.

References AST_LIST_EMPTY, ast_string_field_set, ast_strlen_zero(), check_sip_domain(), get_domain(), get_header(), MAXHOSTNAMELEN, and sip_cfg.

Referenced by transmit_response_with_auth().

{
   char domain[MAXHOSTNAMELEN];

   if (!ast_strlen_zero(p->realm))
      return;

   if (sip_cfg.domainsasrealm &&
       !AST_LIST_EMPTY(&domain_list))
   {
      /* Check From header first */
      if (!get_domain(get_header(req, "From"), domain, sizeof(domain))) {
         if (check_sip_domain(domain, NULL, 0)) {
            ast_string_field_set(p, realm, domain);
            return;
         }
      }
      /* Check To header */
      if (!get_domain(get_header(req, "To"), domain, sizeof(domain))) {
         if (check_sip_domain(domain, NULL, 0)) {
            ast_string_field_set(p, realm, domain);
            return;
         }
      }
   }
   
   /* Use default realm from config file */
   ast_string_field_set(p, realm, sip_cfg.realm);
}

static int get_refer_info	(	struct sip_pvt *	transferer,
		struct sip_request *	outgoing_req
	)			[static]

Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.

Note:

If we get a SIPS uri in the refer-to header, we're required to set up a secure signalling path to that extension. As a minimum, this needs to be added to a channel variable, if not a channel flag.

Definition at line 14445 of file chan_sip.c.

References ast_bridged_channel(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_log(), ast_strdupa, ast_strlen_zero(), ast_uri_decode(), ast_verbose, ast_channel::context, get_header(), LOG_WARNING, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), S_OR, sip_cfg, sip_debug_test_pvt(), SIP_PEDANTIC_DECODE, and strcasestr().

Referenced by handle_request_refer().

{

   const char *p_referred_by = NULL;
   char *h_refer_to = NULL;
   char *h_referred_by = NULL;
   char *refer_to;
   const char *p_refer_to;
   char *referred_by_uri = NULL;
   char *ptr;
   struct sip_request *req = NULL;
   const char *transfer_context = NULL;
   struct sip_refer *referdata;


   req = outgoing_req;
   referdata = transferer->refer;

   if (!req) {
      req = &transferer->initreq;
   }

   p_refer_to = get_header(req, "Refer-To");
   if (ast_strlen_zero(p_refer_to)) {
      ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n");
      return -2;  /* Syntax error */
   }
   h_refer_to = ast_strdupa(p_refer_to);
   refer_to = get_in_brackets(h_refer_to);
   if (!strncasecmp(refer_to, "sip:", 4)) {
      refer_to += 4;       /* Skip sip: */
   } else if (!strncasecmp(refer_to, "sips:", 5)) {
      refer_to += 5;
   } else {
      ast_log(LOG_WARNING, "Can't transfer to non-sip: URI.  (Refer-to: %s)?\n", refer_to);
      return -3;
   }

   /* Get referred by header if it exists */
   p_referred_by = get_header(req, "Referred-By");

   /* Give useful transfer information to the dialplan */
   if (transferer->owner) {
      struct ast_channel *peer = ast_bridged_channel(transferer->owner);
      if (peer) {
         pbx_builtin_setvar_helper(peer, "SIPREFERRINGCONTEXT", transferer->context);
         pbx_builtin_setvar_helper(peer, "SIPREFERREDBYHDR", p_referred_by);
      }
   }

   if (!ast_strlen_zero(p_referred_by)) {
      char *lessthan;
      h_referred_by = ast_strdupa(p_referred_by);

      /* Store referrer's caller ID name */
      ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name));
      if ((lessthan = strchr(referdata->referred_by_name, '<'))) {
         *(lessthan - 1) = '\0'; /* Space */
      }

      referred_by_uri = get_in_brackets(h_referred_by);

      if (!strncasecmp(referred_by_uri, "sip:", 4)) {
         referred_by_uri += 4;      /* Skip sip: */
      } else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
         referred_by_uri += 5;      /* Skip sips: */
      } else {
         ast_log(LOG_WARNING, "Huh?  Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
         referred_by_uri = NULL;
      }
   }

   /* Check for arguments in the refer_to header */
   if ((ptr = strcasestr(refer_to, "replaces="))) {
      char *to = NULL, *from = NULL;
      
      /* This is an attended transfer */
      referdata->attendedtransfer = 1;
      ast_copy_string(referdata->replaces_callid, ptr+9, sizeof(referdata->replaces_callid));
      ast_uri_decode(referdata->replaces_callid);
      if ((ptr = strchr(referdata->replaces_callid, ';')))  /* Find options */ {
         *ptr++ = '\0';
      }
      
      if (ptr) {
         /* Find the different tags before we destroy the string */
         to = strcasestr(ptr, "to-tag=");
         from = strcasestr(ptr, "from-tag=");
      }
      
      /* Grab the to header */
      if (to) {
         ptr = to + 7;
         if ((to = strchr(ptr, '&'))) {
            *to = '\0';
         }
         if ((to = strchr(ptr, ';'))) {
            *to = '\0';
         }
         ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag));
      }
      
      if (from) {
         ptr = from + 9;
         if ((to = strchr(ptr, '&'))) {
            *to = '\0';
         }
         if ((to = strchr(ptr, ';'))) {
            *to = '\0';
         }
         ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
      }
      
      if (!sip_cfg.pedanticsipchecking) {
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
      } else {
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
      }
   }
   
   if ((ptr = strchr(refer_to, '@'))) {   /* Separate domain */
      char *urioption = NULL, *domain;
      int bracket = 0;
      *ptr++ = '\0';

      if ((urioption = strchr(ptr, ';'))) { /* Separate urioptions */
         *urioption++ = '\0';
      }

      domain = ptr;

      /* Remove :port */
      for (; *ptr != '\0'; ++ptr) {
         if (*ptr == ':' && bracket == 0) {
            *ptr = '\0';
            break;
         } else if (*ptr == '[') {
            ++bracket;
         } else if (*ptr == ']') {
            --bracket;
         }
      }

      SIP_PEDANTIC_DECODE(domain);
      SIP_PEDANTIC_DECODE(urioption);

      /* Save the domain for the dial plan */
      ast_copy_string(referdata->refer_to_domain, domain, sizeof(referdata->refer_to_domain));
      if (urioption) {
         ast_copy_string(referdata->refer_to_urioption, urioption, sizeof(referdata->refer_to_urioption));
      }
   }

   if ((ptr = strchr(refer_to, ';')))  /* Remove options */
      *ptr = '\0';

   SIP_PEDANTIC_DECODE(refer_to);
   ast_copy_string(referdata->refer_to, refer_to, sizeof(referdata->refer_to));
   
   if (referred_by_uri) {
      if ((ptr = strchr(referred_by_uri, ';')))    /* Remove options */
         *ptr = '\0';
      SIP_PEDANTIC_DECODE(referred_by_uri);
      ast_copy_string(referdata->referred_by, referred_by_uri, sizeof(referdata->referred_by));
   } else {
      referdata->referred_by[0] = '\0';
   }

   /* Determine transfer context */
   if (transferer->owner)  /* Mimic behaviour in res_features.c */
      transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");

   /* By default, use the context in the channel sending the REFER */
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(transferer->owner->macrocontext,
               S_OR(transferer->context, sip_cfg.default_context));
   }

   ast_copy_string(referdata->refer_to_context, transfer_context, sizeof(referdata->refer_to_context));
   
   /* Either an existing extension or the parking extension */
   if (referdata->attendedtransfer || ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL) ) {
      if (sip_debug_test_pvt(transferer)) {
         ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri);
      }
      /* We are ready to transfer to the extension */
      return 0;
   }
   if (sip_debug_test_pvt(transferer))
      ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context);

   /* Failure, we can't find this extension */
   return -1;
}

static int get_rpid	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)			[static]

Get name, number and presentation from remote party id header, returns true if a valid header was found and it was different from the current caller id.

Definition at line 14039 of file chan_sip.c.

References ast_copy_string(), ast_is_shrinkable_phonenumber(), AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, ast_set_callerid(), ast_shrink_phone_number(), ast_skip_blanks(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, cid_name, cid_num, get_header(), and get_pai().

Referenced by check_peer_ok(), check_user_full(), handle_request_invite(), handle_request_update(), and handle_response_invite().

{
   char tmp[256];
   struct sip_request *req;
   char *cid_num = "";
   char *cid_name = "";
   int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
   char *privacy = "";
   char *screen = "";
   char *start, *end;

   if (!ast_test_flag(&p->flags[0], SIP_TRUSTRPID))
      return 0;
   req = oreq;
   if (!req)
      req = &p->initreq;
   ast_copy_string(tmp, get_header(req, "Remote-Party-ID"), sizeof(tmp));
   if (ast_strlen_zero(tmp)) {
      return get_pai(p, req);
   }

   start = tmp;
   if (*start == '"') {
      *start++ = '\0';
      end = strchr(start, '"');
      if (!end)
         return 0;
      *end++ = '\0';
      cid_name = start;
      start = ast_skip_blanks(end);
   }

   if (*start != '<')
      return 0;
   *start++ = '\0';
   end = strchr(start, '@');
   if (!end)
      return 0;
   *end++ = '\0';
   if (strncasecmp(start, "sip:", 4))
      return 0;
   cid_num = start + 4;
   if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num))
      ast_shrink_phone_number(cid_num);
   start = end;

   end = strchr(start, '>');
   if (!end)
      return 0;
   *end++ = '\0';
   if (*end) {
      start = end;
      if (*start != ';')
         return 0;
      *start++ = '\0';
      while (!ast_strlen_zero(start)) {
         end = strchr(start, ';');
         if (end)
            *end++ = '\0';
         if (!strncasecmp(start, "privacy=", 8))
            privacy = start + 8;
         else if (!strncasecmp(start, "screen=", 7))
            screen = start + 7;
         start = end;
      }

      if (!strcasecmp(privacy, "full")) {
         if (!strcasecmp(screen, "yes"))
            callingpres = AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN;
         else if (!strcasecmp(screen, "no"))
            callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
      } else {
         if (!strcasecmp(screen, "yes"))
            callingpres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
         else if (!strcasecmp(screen, "no"))
            callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
      }
   }

   /* Only return true if the supplied caller id is different */
   if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres)
      return 0;

   ast_string_field_set(p, cid_num, cid_num);
   ast_string_field_set(p, cid_name, cid_name);
   p->callingpres = callingpres;

   if (p->owner) {
      ast_set_callerid(p->owner, cid_num, cid_name, NULL);
      p->owner->caller.id.name.presentation = callingpres;
      p->owner->caller.id.number.presentation = callingpres;
   }

   return 1;
}

static const char * get_sdp_iterate	(	int *	start,
		struct sip_request *	req,
		const char *	name
	)			[static]

Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.

Definition at line 6681 of file chan_sip.c.

References get_body_by_line(), and len().

{
   int len = strlen(name);

   while (*start < (req->sdp_start + req->sdp_count)) {
      const char *r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[(*start)++]), name, len, '=');
      if (r[0] != '\0')
         return r;
   }

   /* if the line was not found, ensure that *start points past the SDP */
   (*start)++;

   return "";
}

static char get_sdp_line	(	int *	start,
		int	stop,
		struct sip_request *	req,
		const char **	value
	)			[static]

Fetches the next valid SDP line between the 'start' line (inclusive) and the 'stop' line (exclusive). Returns the type ('a', 'c', ...) and matching line in reference 'start' is updated with the next line number.

Definition at line 6702 of file chan_sip.c.

References ast_skip_blanks(), and type.

Referenced by process_sdp().

{
   char type = '\0';
   const char *line = NULL;

   if (stop > (req->sdp_start + req->sdp_count)) {
      stop = req->sdp_start + req->sdp_count;
   }

   while (*start < stop) {
      line = REQ_OFFSET_TO_STR(req, line[(*start)++]);
      if (line[1] == '=') {
         type = line[0];
         *value = ast_skip_blanks(line + 2);
         break;
      }
   }

   return type;
}

static struct sip_pvt * get_sip_pvt_byid_locked	(	const char *	callid,
		const char *	totag,
		const char *	fromtag
	)			[static, read]

Lock dialog lock and find matching pvt lock.

Returns:

a reference, remember to release it when done

Definition at line 14356 of file chan_sip.c.

References ao2_t_find, ast_channel_trylock, ast_debug, ast_strlen_zero(), OBJ_POINTER, sip_cfg, sip_pvt_lock, sip_pvt_unlock, and TRUE.

Referenced by handle_request_invite(), and local_attended_transfer().

{
   struct sip_pvt *sip_pvt_ptr;
   struct sip_pvt tmp_dialog = {
      .callid = callid,
   };

   if (totag) {
      ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>");
   }

   /* Search dialogs and find the match */
   
   sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
   if (sip_pvt_ptr) {
      /* Go ahead and lock it (and its owner) before returning */
      sip_pvt_lock(sip_pvt_ptr);
      if (sip_cfg.pedanticsipchecking) {
         unsigned char frommismatch = 0, tomismatch = 0;

         if (ast_strlen_zero(fromtag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }

         if (ast_strlen_zero(totag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }
         /* RFC 3891
          * > 3.  User Agent Server Behavior: Receiving a Replaces Header
          * > The Replaces header contains information used to match an existing
          * > SIP dialog (call-id, to-tag, and from-tag).  Upon receiving an INVITE
          * > with a Replaces header, the User Agent (UA) attempts to match this
          * > information with a confirmed or early dialog.  The User Agent Server
          * > (UAS) matches the to-tag and from-tag parameters as if they were tags
          * > present in an incoming request.  In other words, the to-tag parameter
          * > is compared to the local tag, and the from-tag parameter is compared
          * > to the remote tag.
          *
          * Thus, the totag is always compared to the local tag, regardless if
          * this our call is an incoming or outgoing call.
          */
         frommismatch = !!strcmp(fromtag, sip_pvt_ptr->theirtag);
         tomismatch = !!strcmp(totag, sip_pvt_ptr->tag);

         if (frommismatch || tomismatch) {
            sip_pvt_unlock(sip_pvt_ptr);
            if (frommismatch) {
               ast_debug(4, "Matched %s call for callid=%s - pedantic from tag check fails; their tag is %s our tag is %s\n",
                    sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
                    fromtag, sip_pvt_ptr->theirtag);
            }
            if (tomismatch) {
               ast_debug(4, "Matched %s call for callid=%s - pedantic to tag check fails; their tag is %s our tag is %s\n",
                    sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
                    totag, sip_pvt_ptr->tag);
            }
            return NULL;
         }
      }
      
      if (totag)
         ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
                 sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);

      /* deadlock avoidance... */
      while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) {
         sip_pvt_unlock(sip_pvt_ptr);
         usleep(1);
         sip_pvt_lock(sip_pvt_ptr);
      }
   }
   
   return sip_pvt_ptr;
}

static const char* get_srv_protocol

(

enum sip_transport

t

)

[inline, static]

Return protocol string for srv dns query.

Definition at line 3034 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do(), build_peer(), create_addr(), and transmit_register().

{
   switch (t) {
   case SIP_TRANSPORT_UDP:
      return "udp";
   case SIP_TRANSPORT_TLS:
   case SIP_TRANSPORT_TCP:
      return "tcp";
   }

   return "udp";
}

static const char* get_srv_service

(

enum sip_transport

t

)

[inline, static]

Return service string for srv dns query.

Definition at line 3048 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do(), build_peer(), create_addr(), and transmit_register().

{
   switch (t) {
   case SIP_TRANSPORT_TCP:
   case SIP_TRANSPORT_UDP:
      return "sip";
   case SIP_TRANSPORT_TLS:
      return "sips";
   }
   return "sip";
}

static const char* get_transport

(

enum sip_transport

t

)

[inline, static]

Return transport as string.

Definition at line 3019 of file chan_sip.c.

Referenced by _sip_show_peer(), ast_sip_ouraddrfor(), build_contact(), get_transport_pvt(), handle_request_do(), parse_moved_contact(), sip_show_settings(), sip_show_tcp(), and transmit_notify_with_mwi().

{
   switch (t) {
   case SIP_TRANSPORT_UDP:
      return "UDP";
   case SIP_TRANSPORT_TCP:
      return "TCP";
   case SIP_TRANSPORT_TLS:
      return "TLS";
   }

   return "UNKNOWN";
}

static const char* get_transport_list

(

unsigned int

transports

)

[inline, static]

Return configuration of transports for a device.

Definition at line 2998 of file chan_sip.c.

Referenced by _sip_show_peer(), peers_data_provider_get(), and sip_show_settings().

                                                                      {
   switch (transports) {
      case SIP_TRANSPORT_UDP:
         return "UDP";
      case SIP_TRANSPORT_TCP:
         return "TCP";
      case SIP_TRANSPORT_TLS:
         return "TLS";
      case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP:
         return "TCP,UDP";
      case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TLS:
         return "TLS,UDP";
      case SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS:
         return "TLS,TCP";
      default:
         return transports ?
            "TLS,TCP,UDP" : "UNKNOWN"; 
   }
}

static const char* get_transport_pvt

(

struct sip_pvt *

p

)

[inline, static]

Return transport of dialog.

Note:

this is based on a false assumption. We don't always use the outbound proxy for all requests in a dialog. It depends on the "force" parameter. The FIRST request is always sent to the ob proxy.

Todo:

Fix this function to work correctly

Definition at line 3066 of file chan_sip.c.

References get_transport(), and set_socket_transport().

Referenced by __sip_xmit(), and build_via().

{
   if (p->outboundproxy && p->outboundproxy->transport) {
      set_socket_transport(&p->socket, p->outboundproxy->transport);
   }

   return get_transport(p->socket.type);
}

static int get_transport_str2enum

(

const char *

transport

)

[static]

Return int representing a bit field of transport types found in const char *transport.

Definition at line 2976 of file chan_sip.c.

References ast_strlen_zero().

Referenced by __set_address_from_contact(), and parse_register_contact().

{
   int res = 0;

   if (ast_strlen_zero(transport)) {
      return res;
   }

   if (!strcasecmp(transport, "udp")) {
      res |= SIP_TRANSPORT_UDP;
   }
   if (!strcasecmp(transport, "tcp")) {
      res |= SIP_TRANSPORT_TCP;
   }
   if (!strcasecmp(transport, "tls")) {
      res |= SIP_TRANSPORT_TLS;
   }

   return res;
}

static const char * gettag	(	const struct sip_request *	req,
		const char *	header,
		char *	tagbuf,
		int	tagbufsize
	)			[static]

Get tag from packet.

Returns:

Returns the pointer to the provided tag buffer, or NULL if the tag was not found.

Definition at line 20251 of file chan_sip.c.

References ast_copy_string(), get_header(), strcasestr(), and strsep().

Referenced by find_call(), handle_incoming(), handle_request_subscribe(), and handle_response().

{
   const char *thetag;

   if (!tagbuf)
      return NULL;
   tagbuf[0] = '\0';    /* reset the buffer */
   thetag = get_header(req, header);
   thetag = strcasestr(thetag, ";tag=");
   if (thetag) {
      thetag += 5;
      ast_copy_string(tagbuf, thetag, tagbufsize);
      return strsep(&tagbuf, ";");
   }
   return NULL;
}

static int handle_cc_notify	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)			[static]

Definition at line 20268 of file chan_sip.c.

References ao2_callback, ao2_ref, ast_cc_monitor_callee_available(), ast_cc_monitor_request_acked(), ast_string_field_set, ast_strlen_zero(), construct_pidf_body(), find_sip_monitor_instance_by_subscription_pvt(), get_body(), get_header(), status, transmit_publish(), and transmit_response().

Referenced by handle_request_notify().

{
   struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
         find_sip_monitor_instance_by_subscription_pvt, pvt);
   const char *status = get_body(req, "cc-state", ':');
   struct cc_epa_entry *cc_entry;
   char *uri;

   if (!monitor_instance) {
      transmit_response(pvt, "400 Bad Request", req);
      return -1;
   }

   if (ast_strlen_zero(status)) {
      ao2_ref(monitor_instance, -1);
      transmit_response(pvt, "400 Bad Request", req);
      return -1;
   }

   if (!strcmp(status, "queued")) {
      /* We've been told that we're queued. This is the endpoint's way of telling
       * us that it has accepted our CC request. We need to alert the core of this
       * development
       */
      ast_cc_monitor_request_acked(monitor_instance->core_id, "SIP endpoint %s accepted request", monitor_instance->device_name);
      transmit_response(pvt, "200 OK", req);
      ao2_ref(monitor_instance, -1);
      return 0;
   }

   /* It's open! Yay! */
   uri = get_body(req, "cc-URI", ':');
   if (ast_strlen_zero(uri)) {
      uri = get_in_brackets((char *)get_header(req, "From"));
   }

   ast_string_field_set(monitor_instance, notify_uri, uri);
   if (monitor_instance->suspension_entry) {
      cc_entry = monitor_instance->suspension_entry->instance_data;
      if (cc_entry->current_state == CC_CLOSED) {
         /* If we've created a suspension entry and the current state is closed, then that means
          * we got a notice from the CC core earlier to suspend monitoring, but because this particular
          * call leg had not yet notified us that it was ready for recall, it meant that we
          * could not yet send a PUBLISH. Now, however, we can.
          */
         construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body,
               sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
         transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_INITIAL, monitor_instance->notify_uri);
      } else {
         ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
      }
   } else {
      ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
   }
   ao2_ref(monitor_instance, -1);
   transmit_response(pvt, "200 OK", req);

   return 0;
}

static int handle_cc_subscribe	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Definition at line 22895 of file chan_sip.c.

References ao2_ref, ast_cc_agent_accept_request(), ast_cc_failed(), ast_log(), ast_strlen_zero(), ast_cc_agent::core_id, ast_cc_agent::device_name, find_sip_cc_agent_by_subscribe_uri(), get_header(), LOG_WARNING, ast_cc_agent::private_data, and transmit_response().

Referenced by handle_request_subscribe().

{
   const char *uri = REQ_OFFSET_TO_STR(req, rlPart2);
   char *param_separator;
   struct ast_cc_agent *agent;
   struct sip_cc_agent_pvt *agent_pvt;
   const char *expires_str = get_header(req, "Expires");
   int expires = -1; /* Just need it to be non-zero */

   if (!ast_strlen_zero(expires_str)) {
      sscanf(expires_str, "%d", &expires);
   }

   if ((param_separator = strchr(uri, ';'))) {
      *param_separator = '\0';
   }

   if (!(agent = find_sip_cc_agent_by_subscribe_uri(uri))) {
      if (!expires) {
         /* Typically, if a 0 Expires reaches us and we can't find
          * the corresponding agent, it means that the CC transaction
          * has completed and so the calling side is just trying to
          * clean up its subscription. We'll just respond with a
          * 200 OK and be done with it
          */
         transmit_response(p, "200 OK", req);
         return 0;
      }
      ast_log(LOG_WARNING, "Invalid URI '%s' in CC subscribe\n", uri);
      transmit_response(p, "404 Not Found", req);
      return -1;
   }

   agent_pvt = agent->private_data;

   if (!expires) {
      /* We got sent a SUBSCRIBE and found an agent. This means that CC
       * is being canceled.
       */
      ast_cc_failed(agent->core_id, "CC is being canceled by %s", agent->device_name);
      transmit_response(p, "200 OK", req);
      ao2_ref(agent, -1);
      return 0;
   }

   agent_pvt->subscribe_pvt = dialog_ref(p, "SIP CC agent gains reference to subscription dialog");
   ast_cc_agent_accept_request(agent->core_id, "SIP caller %s has requested CC via SUBSCRIBE",
         agent->device_name);
   p->subscribed = CALL_COMPLETION;

   /* We don't send a response here. That is done in the agent's ack callback or in the
    * agent destructor, should a failure occur before we have responded
    */
   ao2_ref(agent, -1);
   return 0;
}

static int handle_common_options	(	struct ast_flags *	flags,
		struct ast_flags *	mask,
		struct ast_variable *	v
	)			[static]

Handle flag-type options common to configuration of devices - peers.

Parameters:

	flags	array of two struct ast_flags
	mask	array of two struct ast_flags
	v	linked list of config variables to process

Returns:

non-zero if any config options were handled, zero otherwise

Definition at line 25136 of file chan_sip.c.

References ast_clear_flag, ast_copy_string(), ast_false(), ast_log(), ast_set2_flag, ast_set_flag, ast_strdupa, ast_true(), ast_variable::lineno, LOG_WARNING, ast_variable::name, set_insecure_flags(), strsep(), ast_variable::value, and word.

Referenced by build_peer(), and reload_config().

{
   int res = 1;

   if (!strcasecmp(v->name, "trustrpid")) {
      ast_set_flag(&mask[0], SIP_TRUSTRPID);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID);
   } else if (!strcasecmp(v->name, "sendrpid")) {
      ast_set_flag(&mask[0], SIP_SENDRPID);
      if (!strcasecmp(v->value, "pai")) {
         ast_set_flag(&flags[0], SIP_SENDRPID_PAI);
      } else if (!strcasecmp(v->value, "rpid")) {
         ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
      } else if (ast_true(v->value)) {
         ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
      }
   } else if (!strcasecmp(v->name, "rpid_update")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RPID_UPDATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_UPDATE);
   } else if (!strcasecmp(v->name, "rpid_immediate")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RPID_IMMEDIATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_IMMEDIATE);
   } else if (!strcasecmp(v->name, "g726nonstandard")) {
      ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
   } else if (!strcasecmp(v->name, "useclientcode")) {
      ast_set_flag(&mask[0], SIP_USECLIENTCODE);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE);
   } else if (!strcasecmp(v->name, "dtmfmode")) {
      ast_set_flag(&mask[0], SIP_DTMF);
      ast_clear_flag(&flags[0], SIP_DTMF);
      if (!strcasecmp(v->value, "inband"))
         ast_set_flag(&flags[0], SIP_DTMF_INBAND);
      else if (!strcasecmp(v->value, "rfc2833"))
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      else if (!strcasecmp(v->value, "info"))
         ast_set_flag(&flags[0], SIP_DTMF_INFO);
      else if (!strcasecmp(v->value, "shortinfo"))
         ast_set_flag(&flags[0], SIP_DTMF_SHORTINFO);
      else if (!strcasecmp(v->value, "auto"))
         ast_set_flag(&flags[0], SIP_DTMF_AUTO);
      else {
         ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno);
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      }
   } else if (!strcasecmp(v->name, "nat")) {
      ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
      if (!strcasecmp(v->value, "no")) {
         ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
      } else if (!strcasecmp(v->value, "force_rport")) {
         ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
      } else if (!strcasecmp(v->value, "yes")) {
         ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
         ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
         ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
      } else if (!strcasecmp(v->value, "comedia")) {
         ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
         ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
         ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
      }
   } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
      ast_set_flag(&mask[0], SIP_REINVITE);
      ast_clear_flag(&flags[0], SIP_REINVITE);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT);
      } else if (!ast_false(v->value)) {
         char buf[64];
         char *word, *next = buf;

         ast_copy_string(buf, v->value, sizeof(buf));
         while ((word = strsep(&next, ","))) {
            if (!strcasecmp(word, "update")) {
               ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA);
            } else if (!strcasecmp(word, "nonat")) {
               ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
               ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
            } else {
               ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
            }
         }
      }
   } else if (!strcasecmp(v->name, "insecure")) {
      ast_set_flag(&mask[0], SIP_INSECURE);
      ast_clear_flag(&flags[0], SIP_INSECURE);
      set_insecure_flags(&flags[0], v->value, v->lineno);   
   } else if (!strcasecmp(v->name, "progressinband")) {
      ast_set_flag(&mask[0], SIP_PROG_INBAND);
      ast_clear_flag(&flags[0], SIP_PROG_INBAND);
      if (ast_true(v->value))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_YES);
      else if (strcasecmp(v->value, "never"))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_NO);
   } else if (!strcasecmp(v->name, "promiscredir")) {
      ast_set_flag(&mask[0], SIP_PROMISCREDIR);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR);
   } else if (!strcasecmp(v->name, "videosupport")) {
      if (!strcasecmp(v->value, "always")) {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
         ast_set_flag(&flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
      } else {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
         ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
      }
   } else if (!strcasecmp(v->name, "textsupport")) {
      ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
      res = 1;
   } else if (!strcasecmp(v->name, "allowoverlap")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
   } else if (!strcasecmp(v->name, "allowsubscribe")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
   } else if (!strcasecmp(v->name, "ignoresdpversion")) {
      ast_set_flag(&mask[1], SIP_PAGE2_IGNORESDPVERSION);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_IGNORESDPVERSION);
   } else if (!strcasecmp(v->name, "faxdetect")) {
      ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
      } else if (ast_false(v->value)) {
         ast_clear_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
      } else {
         char *buf = ast_strdupa(v->value);
         char *word, *next = buf;

         while ((word = strsep(&next, ","))) {
            if (!strcasecmp(word, "cng")) {
               ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_CNG);
            } else if (!strcasecmp(word, "t38")) {
               ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_T38);
            } else {
               ast_log(LOG_WARNING, "Unknown faxdetect mode '%s' on line %d.\n", word, v->lineno);
            }
         }
      }
   } else if (!strcasecmp(v->name, "rfc2833compensate")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
   } else if (!strcasecmp(v->name, "buggymwi")) {
      ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
   } else
      res = 0;

   return res;
}

static int handle_incoming	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		int *	recount,
		int *	nounlock
	)			[static]

Handle incoming SIP requests (methods).

Note:

This is where all incoming requests go first

Definition at line 23392 of file chan_sip.c.

References __sip_ack(), ast_channel_ref, ast_channel_unlock, ast_channel_unref, ast_debug, ast_log(), ast_skip_blanks(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_verbose, check_pendings(), debug, extract_uri(), find_sdp(), get_header(), gettag(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_message(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_request_update(), handle_response(), cfsip_methods::id, len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, ast_channel::name, pbx_builtin_setvar_helper(), process_sdp(), pvt_set_needdestroy(), sip_cfg, sip_debug_test_pvt(), sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), cfsip_methods::text, transmit_response(), transmit_response_reliable(), and transmit_response_with_allow().

Referenced by handle_request_do(), and process_request_queue().

{
   /* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
      relatively static */
   const char *cmd;
   const char *cseq;
   const char *useragent;
   int seqno;
   int len;
   int respid;
   int res = 0;
   int debug = sip_debug_test_pvt(p);
   const char *e;
   int error = 0;
   int oldmethod = p->method;
   int acked = 0;

   /* Get Method and Cseq */
   cseq = get_header(req, "Cseq");
   cmd = REQ_OFFSET_TO_STR(req, header[0]);

   /* Must have Cseq */
   if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq)) {
      ast_log(LOG_ERROR, "Missing Cseq. Dropping this SIP message, it's incomplete.\n");
      error = 1;
   }
   if (!error && sscanf(cseq, "%30d%n", &seqno, &len) != 1) {
      ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd);
      error = 1;
   }
   if (error) {
      if (!p->initreq.headers) { /* New call */
         pvt_set_needdestroy(p, "no headers");
      }
      return -1;
   }
   /* Get the command XXX */

   cmd = REQ_OFFSET_TO_STR(req, rlPart1);
   e = ast_skip_blanks(REQ_OFFSET_TO_STR(req, rlPart2));

   /* Save useragent of the client */
   useragent = get_header(req, "User-Agent");
   if (!ast_strlen_zero(useragent))
      ast_string_field_set(p, useragent, useragent);

   /* Find out SIP method for incoming request */
   if (req->method == SIP_RESPONSE) {  /* Response to our request */
      /* ignore means "don't do anything with it" but still have to
       * respond appropriately.
       * But in this case this is a response already, so we really
       * have nothing to do with this message, and even setting the
       * ignore flag is pointless.
       */
      if (ast_strlen_zero(e)) {
         return 0;
      }
      if (sscanf(e, "%30d %n", &respid, &len) != 1) {
         ast_log(LOG_WARNING, "Invalid response: '%s'\n", e);
         return 0;
      }
      if (respid <= 0) {
         ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid);
         return 0;
      }
      if (p->ocseq && (p->ocseq < seqno)) {
         ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
         return -1;
      } else {
         char causevar[256], causeval[256];

         if ((respid == 200) || ((respid >= 300) && (respid <= 399))) {
            extract_uri(p, req);
         }

         handle_response(p, respid, e + len, req, seqno);

         if (p->owner) {
            struct ast_channel *owner = p->owner;

            snprintf(causevar, sizeof(causevar), "MASTER_CHANNEL(HASH(SIP_CAUSE,%s))", owner->name);
            snprintf(causeval, sizeof(causeval), "SIP %s", REQ_OFFSET_TO_STR(req, rlPart2));

            ast_channel_ref(owner);
            sip_pvt_unlock(p);
            ast_channel_unlock(owner);
            *nounlock = 1;
            pbx_builtin_setvar_helper(owner, causevar, causeval);
            ast_channel_unref(owner);
            sip_pvt_lock(p);
         }
      }
      return 0;
   }

   /* New SIP request coming in
      (could be new request in existing SIP dialog as well...)
    */         
   
   p->method = req->method;   /* Find out which SIP method they are using */
   ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);

   if (p->icseq && (p->icseq > seqno) ) {
      if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
         ast_debug(2, "Got CANCEL or ACK on INVITE with transactions in between.\n");
      }  else {
         ast_debug(1, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq);
         if (req->method != SIP_ACK)
            transmit_response(p, "503 Server error", req);  /* We must respond according to RFC 3261 sec 12.2 */
         return -1;
      }
   } else if (p->icseq &&
         p->icseq == seqno &&
         req->method != SIP_ACK &&
         (p->method != SIP_CANCEL || p->alreadygone)) {
      /* ignore means "don't do anything with it" but still have to
         respond appropriately.  We do this if we receive a repeat of
         the last sequence number  */
      req->ignore = 1;
      ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
   }

   /* RFC 3261 section 9. "CANCEL has no effect on a request to which a UAS has
    * already given a final response." */
   if (!p->pendinginvite && (req->method == SIP_CANCEL)) {
      transmit_response(p, "481 Call/Transaction Does Not Exist", req);
      return res;
   }

   if (seqno >= p->icseq)
      /* Next should follow monotonically (but not necessarily
         incrementally -- thanks again to the genius authors of SIP --
         increasing */
      p->icseq = seqno;

   /* Find their tag if we haven't got it */
   if (ast_strlen_zero(p->theirtag)) {
      char tag[128];

      gettag(req, "From", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   }
   snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);

   if (sip_cfg.pedanticsipchecking) {
      /* If this is a request packet without a from tag, it's not
         correct according to RFC 3261  */
      /* Check if this a new request in a new dialog with a totag already attached to it,
         RFC 3261 - section 12.2 - and we don't want to mess with recovery  */
      if (!p->initreq.headers && req->has_to_tag) {
         /* If this is a first request and it got a to-tag, it is not for us */
         if (!req->ignore && req->method == SIP_INVITE) {
            transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
            /* Will cease to exist after ACK */
         } else if (req->method != SIP_ACK) {
            transmit_response(p, "481 Call/Transaction Does Not Exist", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         } else {
            ast_debug(1, "Got ACK for unknown dialog... strange.\n");
         }
         return res;
      }
   }

   if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY || p->method == SIP_PUBLISH)) {
      transmit_response(p, "400 Bad request", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return -1;
   }

   /* Handle various incoming SIP methods in requests */
   switch (p->method) {
   case SIP_OPTIONS:
      res = handle_request_options(p, req, addr, e);
      break;
   case SIP_INVITE:
      res = handle_request_invite(p, req, debug, seqno, addr, recount, e, nounlock);
      break;
   case SIP_REFER:
      res = handle_request_refer(p, req, debug, seqno, nounlock);
      break;
   case SIP_CANCEL:
      res = handle_request_cancel(p, req);
      break;
   case SIP_BYE:
      res = handle_request_bye(p, req);
      break;
   case SIP_MESSAGE:
      res = handle_request_message(p, req);
      break;
   case SIP_PUBLISH:
      res = handle_request_publish(p, req, addr, seqno, e);
      break;
   case SIP_SUBSCRIBE:
      res = handle_request_subscribe(p, req, addr, seqno, e);
      break;
   case SIP_REGISTER:
      res = handle_request_register(p, req, addr, e);
      break;
   case SIP_INFO:
      if (req->debug)
         ast_verbose("Receiving INFO!\n");
      if (!req->ignore)
         handle_request_info(p, req);
      else  /* if ignoring, transmit response */
         transmit_response(p, "200 OK", req);
      break;
   case SIP_NOTIFY:
      res = handle_request_notify(p, req, addr, seqno, e);
      break;
   case SIP_UPDATE:
      res = handle_request_update(p, req);
      break;
   case SIP_ACK:
      /* Make sure we don't ignore this */
      if (seqno == p->pendinginvite) {
         p->invitestate = INV_TERMINATED;
         p->pendinginvite = 0;
         acked = __sip_ack(p, seqno, 1 /* response */, 0);
         if (find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_NONE))
               return -1;
         }
         check_pendings(p);
      } else if (p->glareinvite == seqno) {
         /* handle ack for the 491 pending sent for glareinvite */
         p->glareinvite = 0;
         acked = __sip_ack(p, seqno, 1, 0);
      }
      if (!acked) {
         /* Got an ACK that did not match anything. Ignore
          * silently and restore previous method */
         p->method = oldmethod;
      }
      if (!p->lastinvite && ast_strlen_zero(p->randdata)) {
         pvt_set_needdestroy(p, "unmatched ACK");
      }
      break;
   default:
      transmit_response_with_allow(p, "501 Method Not Implemented", req, 0);
      ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n",
         cmd, ast_sockaddr_stringify(&p->sa));
      /* If this is some new method, and we don't have a call, destroy it now */
      if (!p->initreq.headers) {
         pvt_set_needdestroy(p, "unimplemented method");
      }
      break;
   }
   return res;
}

static int handle_invite_replaces	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		struct ast_sockaddr *	addr,
		int *	nounlock
	)			[static]

Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.

Note:

this function is called by handle_request_invite(). Four locks held at the beginning of this function, p, p->owner, p->refer->refer_call and p->refere->refer_call->owner. only p's lock should remain at the end of this function. p's lock as well as the channel p->owner's lock are held by handle_request_do(), we unlock p->owner before the masq. By setting nounlock we are indicating to handle_request_do() that we have already unlocked the owner.

Definition at line 20573 of file chan_sip.c.

References ast_channel::_state, append_history, ast_bridged_channel(), AST_CAUSE_SWITCH_CONGESTION, ast_channel_lock, ast_channel_masquerade(), ast_channel_ref, ast_channel_unlock, ast_channel_unref, ast_debug, ast_do_masquerade(), ast_hangup(), ast_log(), ast_quiet_chan(), ast_set_flag, ast_setstate(), AST_STATE_DOWN, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, FALSE, ast_channel::hangupcause, LOG_ERROR, LOG_NOTICE, LOG_WARNING, ast_channel::name, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), ast_channel::tech_pvt, transmit_response(), transmit_response_reliable(), and transmit_response_with_sdp().

Referenced by handle_request_invite().

{
   int earlyreplace = 0;
   int oneleggedreplace = 0;     /* Call with no bridge, propably IVR or voice message */
   struct ast_channel *c = p->owner;   /* Our incoming call */
   struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
   struct ast_channel *targetcall;     /* The bridge to the take-over target */

   /* Check if we're in ring state */
   if (replacecall->_state == AST_STATE_RING)
      earlyreplace = 1;

   /* Check if we have a bridge */
   if (!(targetcall = ast_bridged_channel(replacecall))) {
      /* We have no bridge */
      if (!earlyreplace) {
         ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name);
         oneleggedreplace = 1;
      }
   }
   if (targetcall && targetcall->_state == AST_STATE_RINGING)
      ast_debug(4, "SIP transfer: Target channel is in ringing state\n");

   if (targetcall)
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name);
   else
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name);

   if (req->ignore) {
      ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n");
      /* We should answer something here. If we are here, the
         call we are replacing exists, so an accepted
         can't harm */
      transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
      /* Do something more clever here */
      if (c) {
         *nounlock = 1;
         ast_channel_unlock(c);
      }
      ast_channel_unlock(replacecall);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   }
   if (!c) {
      /* What to do if no channel ??? */
      ast_log(LOG_ERROR, "Unable to create new channel.  Invite/replace failed.\n");
      transmit_response_reliable(p, "503 Service Unavailable", req);
      append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_channel_unlock(replacecall);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   }
   append_history(p, "Xfer", "INVITE/Replace received");
   /* We have three channels to play with
      channel c: New incoming call
      targetcall: Call from PBX to target
      p->refer->refer_call: SIP pvt dialog from transferer to pbx.
      replacecall: The owner of the previous
      We need to masq C into refer_call to connect to
      targetcall;
      If we are talking to internal audio stream, target call is null.
   */

   /* Fake call progress */
   transmit_response(p, "100 Trying", req);
   ast_setstate(c, AST_STATE_RING);

   /* Masquerade the new call into the referred call to connect to target call
      Targetcall is not touched by the masq */

   /* Answer the incoming call and set channel to UP state */
   transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);

   ast_setstate(c, AST_STATE_UP);

   /* Stop music on hold and other generators */
   ast_quiet_chan(replacecall);
   ast_quiet_chan(targetcall);
   ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name);

   /* Make sure that the masq does not free our PVT for the old call */
   if (! earlyreplace && ! oneleggedreplace )
      ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER);  /* Delay hangup */

   /* Prepare the masquerade - if this does not happen, we will be gone */
   if(ast_channel_masquerade(replacecall, c))
      ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n");
   else
      ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name);

   /* C should now be in place of replacecall. all channel locks and pvt locks should be removed
    * before issuing the masq.  Since we are unlocking both the pvt (p) and its owner channel (c)
    * it is possible for channel c to be destroyed on us.  To prevent this, we must give c a reference
    * before any unlocking takes place and remove it only once we are completely done with it */
   ast_channel_ref(c);
   ast_channel_unlock(replacecall);
   ast_channel_unlock(c);
   sip_pvt_unlock(p->refer->refer_call);
   sip_pvt_unlock(p);
   if (ast_do_masquerade(replacecall)) {
      ast_log(LOG_WARNING, "Failed to perform masquerade with INVITE replaces\n");
   }
   ast_channel_lock(c);
   if (earlyreplace || oneleggedreplace ) {
      c->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
   }
   ast_setstate(c, AST_STATE_DOWN);
   ast_channel_unlock(c);

   /* The call should be down with no ast_channel, so hang it up */
   c->tech_pvt = dialog_unref(c->tech_pvt, "unref dialog c->tech_pvt");

   /* c and c's tech pvt must be unlocked at this point for ast_hangup */
   ast_hangup(c);
   /* this indicates to handle_request_do that the owner channel has already been unlocked */
   *nounlock = 1;
   /* lock PVT structure again after hangup */
   sip_pvt_lock(p);
   ast_channel_unref(c);
   return 0;
}

static int handle_request_bye	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming BYE request.

Definition at line 22250 of file chan_sip.c.

References __sip_pretend_ack(), append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_PROTOCOL_ERROR, ast_channel_lock, ast_channel_trylock, ast_channel_unlock, ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, ast_log(), AST_MAX_USER_FIELD, ast_queue_control(), ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_rtp_instance_get_quality(), ast_rtp_instance_set_stats_vars(), AST_RTP_INSTANCE_STAT_FIELD_QUALITY, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, ast_set_hangupsource(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, check_via(), ast_channel::context, context, copy_request(), get_also_info(), get_header(), LOG_NOTICE, LOG_WARNING, pbx_builtin_setvar_helper(), quality, sip_alreadygone(), sip_cfg, sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy_final(), stop_media_flows(), stop_session_timer(), ast_channel::tech, ast_channel::tech_pvt, cfsip_methods::text, transmit_response(), and transmit_response_reliable().

Referenced by handle_incoming().

{
   struct ast_channel *c=NULL;
   int res;
   struct ast_channel *bridged_to;
   
   /* If we have an INCOMING invite that we haven't answered, terminate that transaction */
   if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
   }

   __sip_pretend_ack(p);

   p->invitestate = INV_TERMINATED;

   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);
   sip_alreadygone(p);

   /* Get RTCP quality before end of call */
   if (p->do_history || p->owner) {
      char quality_buf[AST_MAX_USER_FIELD], *quality;
      struct ast_channel *bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;

      /* We need to get the lock on bridge because ast_rtp_instance_set_stats_vars will attempt
       * to lock the bridge. This may get hairy...
       */
      while (bridge && ast_channel_trylock(bridge)) {
         ast_channel_unlock(p->owner);
         do {
            /* Can't use DEADLOCK_AVOIDANCE since p is an ao2 object */
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         } while (p->owner && ast_channel_trylock(p->owner));
         bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;
      }


      if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPaudio", "Quality:%s", quality);

            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioJitter", "Quality:%s", quality);
            }
            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioLoss", "Quality:%s", quality);
            }
            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioRTT", "Quality:%s", quality);
            }
         }

         if (p->owner) {
            ast_rtp_instance_set_stats_vars(p->owner, p->rtp);
         }

      }

      if (bridge) {
         struct sip_pvt *q = bridge->tech_pvt;

         if (IS_SIP_TECH(bridge->tech) && q && q->rtp) {
            ast_rtp_instance_set_stats_vars(bridge, q->rtp);
         }
         ast_channel_unlock(bridge);
      }

      if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPvideo", "Quality:%s", quality);
         }
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
         }
      }
      if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPtext", "Quality:%s", quality);
         }
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
         }
      }
   }

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   stop_session_timer(p); /* Stop Session-Timer */

   if (!ast_strlen_zero(get_header(req, "Also"))) {
      ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method.  Ask vendor to support REFER instead\n",
         ast_sockaddr_stringify(&p->recv));
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, sip_cfg.default_context);
      res = get_also_info(p, req);
      if (!res) {
         c = p->owner;
         if (c) {
            bridged_to = ast_bridged_channel(c);
            if (bridged_to) {
               /* Don't actually hangup here... */
               ast_queue_control(c, AST_CONTROL_UNHOLD);
               ast_channel_unlock(c);  /* async_goto can do a masquerade, no locks can be held during a masq */
               ast_async_goto(bridged_to, p->context, p->refer->refer_to, 1);
               ast_channel_lock(c);
            } else
               ast_queue_hangup(p->owner);
         }
      } else {
         ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_sockaddr_stringify(&p->recv));
         if (p->owner)
            ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
      }
   } else if (p->owner) {
      ast_set_hangupsource(p->owner, p->owner->name, 0);
      ast_queue_hangup(p->owner);
      sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
      ast_debug(3, "Received bye, issuing owner hangup\n");
   } else {
      sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
      ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
   }
   ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   transmit_response(p, "200 OK", req);

   return 1;
}

static int handle_request_cancel	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming CANCEL request.

Definition at line 22185 of file chan_sip.c.

References __sip_pretend_ack(), ast_debug, ast_free, ast_queue_hangup(), AST_SCHED_DEL, ast_set_hangupsource(), AST_STATE_UP, ast_test_flag, check_via(), sip_alreadygone(), sip_scheddestroy(), stop_media_flows(), transmit_response(), transmit_response_reliable(), UNLINK, and update_call_counter().

Referenced by handle_incoming().

{

   check_via(p, req);
   sip_alreadygone(p);

   if (p->owner && p->owner->_state == AST_STATE_UP) {
      /* This call is up, cancel is ignored, we need a bye */
      transmit_response(p, "200 OK", req);
      ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
      return 0;
   }

   /* At this point, we could have cancelled the invite at the same time
      as the other side sends a CANCEL. Our final reply with error code
      might not have been received by the other side before the CANCEL
      was sent, so let's just give up retransmissions and waiting for
      ACK on our error code. The call is hanging up any way. */
   if (p->invitestate == INV_TERMINATED || p->invitestate == INV_COMPLETED) {
      __sip_pretend_ack(p);
   }
   if (p->invitestate != INV_TERMINATED)
      p->invitestate = INV_CANCELLED;

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD))
      update_call_counter(p, DEC_CALL_LIMIT);

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   if (p->owner) {
      ast_set_hangupsource(p->owner, p->owner->name, 0);
      ast_queue_hangup(p->owner);
   }
   else
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   if (p->initreq.len > 0) {
      struct sip_pkt *pkt, *prev_pkt;
      /* If the CANCEL we are receiving is a retransmission, and we already have scheduled
       * a reliable 487, then we don't want to schedule another one on top of the previous
       * one.
       *
       * As odd as this may sound, we can't rely on the previously-transmitted "reliable"
       * response in this situation. What if we've sent all of our reliable responses
       * already and now all of a sudden, we get this second CANCEL?
       *
       * The only way to do this correctly is to cancel our previously-scheduled reliably-
       * transmitted response and send a new one in its place.
       */
      for (pkt = p->packets, prev_pkt = NULL; pkt; prev_pkt = pkt, pkt = pkt->next) {
         if (pkt->seqno == p->lastinvite && pkt->response_code == 487) {
            AST_SCHED_DEL(sched, pkt->retransid);
            UNLINK(pkt, p->packets, prev_pkt);
            ast_free(pkt);
            break;
         }
      }
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
      transmit_response(p, "200 OK", req);
      return 1;
   } else {
      transmit_response(p, "481 Call Leg Does Not Exist", req);
      return 0;
   }
}

static int handle_request_do	(	struct sip_request *	req,
		struct ast_sockaddr *	addr
	)			[static]

Handle incoming SIP message - request or response.

This is used for all transports (udp, tcp and tcp/tls)

Definition at line 23781 of file chan_sip.c.

References ao2_t_ref, append_history, ast_channel_trylock, ast_channel_unlock, ast_debug, AST_LIST_EMPTY, ast_log(), ast_mutex_lock, ast_mutex_unlock, AST_SCHED_DEL_UNREF, ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_str_reset(), ast_update_use_count(), ast_verbose, copy_socket_data(), find_call(), find_sip_method(), get_header(), get_transport(), handle_incoming(), LOG_ERROR, lws2sws(), netlock, parse_request(), process_request_queue(), queue_request(), S_OR, sip_cfg, sip_debug_test_addr(), sip_pvt_unlock, and transmit_response().

Referenced by _sip_tcp_helper_thread(), and sipsock_read().

{
   struct sip_pvt *p;
   int recount = 0;
   int nounlock = 0;
   int lockretry;

   if (sip_debug_test_addr(addr))   /* Set the debug flag early on packet level */
      req->debug = 1;
   if (sip_cfg.pedanticsipchecking)
      req->len = lws2sws(req->data->str, req->len);   /* Fix multiline headers */
   if (req->debug) {
      ast_verbose("\n<--- SIP read from %s:%s --->\n%s\n<------------->\n",
         get_transport(req->socket.type), ast_sockaddr_stringify(addr), req->data->str);
   }

   if (parse_request(req) == -1) { /* Bad packet, can't parse */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }
   req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlPart1));

   if (req->debug)
      ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");

   if (req->headers < 2) { /* Must have at least two headers */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }

   /* Process request, with netlock held, and with usual deadlock avoidance */
   for (lockretry = 10; lockretry > 0; lockretry--) {
      ast_mutex_lock(&netlock);

      /* Find the active SIP dialog or create a new one */
      p = find_call(req, addr, req->method); /* returns p locked */
      if (p == NULL) {
         ast_debug(1, "Invalid SIP message - rejected , no callid, len %d\n", req->len);
         ast_mutex_unlock(&netlock);
         return 1;
      }

      copy_socket_data(&p->socket, &req->socket);

      /* Go ahead and lock the owner if it has one -- we may need it */
      /* becaues this is deadlock-prone, we need to try and unlock if failed */
      if (!p->owner || !ast_channel_trylock(p->owner))
         break;   /* locking succeeded */

      if (lockretry != 1) {
         sip_pvt_unlock(p);
         ao2_t_ref(p, -1, "release p (from find_call) inside lockretry loop"); /* we'll look for it again, but p is dead now */
         ast_mutex_unlock(&netlock);
         /* Sleep for a very short amount of time */
         usleep(1);
      }
   }
   ast_sockaddr_copy(&p->recv, addr);

   if (p->do_history) /* This is a request or response, note what it was for */
      append_history(p, "Rx", "%s / %s / %s", req->data->str, get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlPart2));

   if (!lockretry) {
      if (!queue_request(p, req)) {
         /* the request has been queued for later handling */
         sip_pvt_unlock(p);
         ao2_t_ref(p, -1, "release p (from find_call) after queueing request");
         ast_mutex_unlock(&netlock);
         return 1;
      }

      if (p->owner)
         ast_log(LOG_ERROR, "Channel lock for %s could not be obtained, and request was unable to be queued.\n", S_OR(p->owner->name, "- no channel name ??? - "));
      ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
      if (req->method != SIP_ACK)
         transmit_response(p, "503 Server error", req);  /* We must respond according to RFC 3261 sec 12.2 */
      /* XXX We could add retry-after to make sure they come back */
      append_history(p, "LockFail", "Owner lock failed, transaction failed.");
      sip_pvt_unlock(p);
      ao2_t_ref(p, -1, "release p (from find_call) at end of lockretry"); /* p is gone after the return */
      ast_mutex_unlock(&netlock);
      return 1;
   }

   /* if there are queued requests on this sip_pvt, process them first, so that everything is
      handled in order
   */
   if (!AST_LIST_EMPTY(&p->request_queue)) {
      AST_SCHED_DEL_UNREF(sched, p->request_queue_sched_id, dialog_unref(p, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
      process_request_queue(p, &recount, &nounlock);
   }

   if (handle_incoming(p, req, addr, &recount, &nounlock) == -1) {
      /* Request failed */
      ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
   }
      
   if (recount)
      ast_update_use_count();

   if (p->owner && !nounlock)
      ast_channel_unlock(p->owner);
   sip_pvt_unlock(p);
   ast_mutex_unlock(&netlock);
   ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
   return 1;
}

static void handle_request_info	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Receive SIP INFO Message.

Todo:

Note: Doesn't read the duration of the DTMF. Should be fixed.

Definition at line 17511 of file chan_sip.c.

References ast_bridged_channel(), ast_cdr_setuserfield(), AST_CONTROL_FLASH, AST_CONTROL_VIDUPDATE, ast_copy_string(), ast_debug, ast_find_call_feature(), AST_FRAME_CONTROL, AST_FRAME_DTMF, ast_log(), ast_queue_control(), ast_queue_frame(), ast_rdlock_call_features(), ast_strlen_zero(), ast_test_flag, ast_unlock_call_features(), ast_verbose, ast_channel::cdr, ast_call_feature::exten, f, get_body(), get_header(), get_msg_text(), ast_frame_subclass::integer, ast_frame::len, LOG_WARNING, sip_scheddestroy(), ast_frame::subclass, transmit_response(), and TRUE.

Referenced by handle_incoming().

{
   char buf[1024];
   unsigned int event;
   const char *c = get_header(req, "Content-Type");

   /* Need to check the media/type */
   if (!strcasecmp(c, "application/dtmf-relay") ||
       !strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
      unsigned int duration = 0;

      if (!p->owner) {  /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      /* Try getting the "signal=" part */
      if (ast_strlen_zero(c = get_body(req, "Signal", '=')) && ast_strlen_zero(c = get_body(req, "d", '='))) {
         ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid);
         transmit_response(p, "200 OK", req); /* Should return error */
         return;
      } else {
         ast_copy_string(buf, c, sizeof(buf));
      }

      if (!ast_strlen_zero((c = get_body(req, "Duration", '=')))) {
         duration = atoi(c);
      }
      if (!duration) {
         duration = 100; /* 100 ms */
      }


      if (ast_strlen_zero(buf)) {
         transmit_response(p, "200 OK", req);
         return;
      }

      if (buf[0] == '*') {
         event = 10;
      } else if (buf[0] == '#') {
         event = 11;
      } else if ((buf[0] >= 'A') && (buf[0] <= 'D')) {
         event = 12 + buf[0] - 'A';
      } else if (buf[0] == '!') {
         event = 16;
      } else {
         event = atoi(buf);
      }
      if (event == 16) {
         /* send a FLASH event */
         struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: FLASH\n");
         }
      } else {
         /* send a DTMF event */
         struct ast_frame f = { AST_FRAME_DTMF, };
         if (event < 10) {
            f.subclass.integer = '0' + event;
         } else if (event == 10) {
            f.subclass.integer = '*';
         } else if (event == 11) {
            f.subclass.integer = '#';
         } else if (event < 16) {
            f.subclass.integer = 'A' + (event - 12);
         }
         f.len = duration;
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
         }
      }
      transmit_response(p, "200 OK", req);
      return;
   } else if (!strcasecmp(c, "application/dtmf")) {
      /*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
      unsigned int duration = 0;

      if (!p->owner) {  /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      get_msg_text(buf, sizeof(buf), req, TRUE);
      duration = 100; /* 100 ms */

      if (ast_strlen_zero(buf)) {
         transmit_response(p, "200 OK", req);
         return;
      }
      event = atoi(buf);
      if (event == 16) {
         /* send a FLASH event */
         struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH }, };
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: FLASH\n");
         }
      } else {
         /* send a DTMF event */
         struct ast_frame f = { AST_FRAME_DTMF, };
         if (event < 10) {
            f.subclass.integer = '0' + event;
         } else if (event == 10) {
            f.subclass.integer = '*';
         } else if (event == 11) {
            f.subclass.integer = '#';
         } else if (event < 16) {
            f.subclass.integer = 'A' + (event - 12);
         }
         f.len = duration;
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
         }
      }
      transmit_response(p, "200 OK", req);
      return;

   } else if (!strcasecmp(c, "application/media_control+xml")) {
      /* Eh, we'll just assume it's a fast picture update for now */
      if (p->owner) {
         ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
      }
      transmit_response(p, "200 OK", req);
      return;
   } else if (!ast_strlen_zero(c = get_header(req, "X-ClientCode"))) {
      /* Client code (from SNOM phone) */
      if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) {
         if (p->owner && p->owner->cdr) {
            ast_cdr_setuserfield(p->owner, c);
         }
         if (p->owner && ast_bridged_channel(p->owner) && ast_bridged_channel(p->owner)->cdr) {
            ast_cdr_setuserfield(ast_bridged_channel(p->owner), c);
         }
         transmit_response(p, "200 OK", req);
      } else {
         transmit_response(p, "403 Forbidden", req);
      }
      return;
   } else if (!ast_strlen_zero(c = get_header(req, "Record"))) {
      /* INFO messages generated by some phones to start/stop recording
         on phone calls.
         OEJ: I think this should be something that is enabled/disabled
         per device. I don't want incoming callers to record calls in my
         pbx.
      */
      /* first, get the feature string, if it exists */
      struct ast_call_feature *feat;
      int j;
      struct ast_frame f = { AST_FRAME_DTMF, };

      ast_rdlock_call_features();
      feat = ast_find_call_feature("automon");
      if (!feat || ast_strlen_zero(feat->exten)) {
         ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
         /* 403 means that we don't support this feature, so don't request it again */
         transmit_response(p, "403 Forbidden", req);
         ast_unlock_call_features();
         return;
      }
      /* Send the feature code to the PBX as DTMF, just like the handset had sent it */
      f.len = 100;
      for (j=0; j < strlen(feat->exten); j++) {
         f.subclass.integer = feat->exten[j];
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event faked: %c\n", f.subclass.integer);
         }
      }
      ast_unlock_call_features();

      ast_debug(1, "Got a Request to Record the channel, state %s\n", c);
      transmit_response(p, "200 OK", req);
      return;
   } else if (ast_strlen_zero(c = get_header(req, "Content-Length")) || !strcasecmp(c, "0")) {
      /* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */
      transmit_response(p, "200 OK", req);
      return;
   }

   /* Other type of INFO message, not really understood by Asterisk */
   /* if (get_msg_text(buf, sizeof(buf), req)) { */

   ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf);
   transmit_response(p, "415 Unsupported media type", req);
   return;
}

static int handle_request_invite	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		struct ast_sockaddr *	addr,
		int *	recount,
		const char *	e,
		int *	nounlock
	)			[static]

Handle incoming INVITE request.

Note:

If the INVITE has a Replaces header, it is part of an attended transfer. If so, we do not go through the dial plan but try to find the active call and masquerade into it

This is a spiral. What we need to do is to just change the outgoing INVITE so that it now routes to the new Request URI. Since we created the INVITE ourselves that should be all we need to do.

Todo:

XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet correctly instead...

Definition at line 20789 of file chan_sip.c.

References __sip_ack(), ast_channel::_state, append_history, ARRAY_LEN, AST_CAUSE_CALL_REJECTED, AST_CAUSE_NORMAL_CLEARING, ast_cc_agent_set_interfaces_chanvar(), ast_channel_lock, ast_channel_queue_connected_line_update(), ast_channel_set_redirecting(), ast_channel_unlock, ast_clear_flag, AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, AST_CONTROL_BUSY, AST_CONTROL_SRCUPDATE, ast_copy_string(), ast_debug, ast_hangup(), ast_log(), AST_MAX_CONTEXT, AST_MAX_EXTENSION, ast_null_frame, ast_party_connected_line_init(), ast_party_redirecting_free(), ast_party_redirecting_init(), AST_PBX_CALL_LIMIT, AST_PBX_FAILED, ast_pbx_start(), AST_PBX_SUCCESS, ast_pickup_call(), ast_pickup_ext(), ast_queue_control(), ast_queue_frame(), ast_rtp_instance_set_alt_remote_address(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_sched_add(), ast_set_flag, ast_setstate(), ast_setup_cc_recall_datastore(), ast_skip_blanks(), AST_STATE_DOWN, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_udptl_new_with_bindaddr(), ast_uri_decode(), ast_verbose, bindaddr, build_contact(), build_route(), change_redirecting_information(), check_user_full(), check_via(), connected, context, copy_request(), do_magic_pickup(), ast_channel::exten, exten, extract_uri(), FALSE, find_sdp(), get_destination(), get_header(), get_ip_and_port_from_sdp(), get_rpid(), get_sip_pvt_byid_locked(), handle_invite_replaces(), ast_channel::hangupcause, ast_party_connected_line::id, ast_set_party_connected_line::id, LOG_ERROR, LOG_NOTICE, LOG_WARNING, make_our_tag(), ast_channel::name, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_minse(), parse_ok_contact(), parse_session_expires(), ast_party_name::presentation, ast_party_number::presentation, process_sdp(), ref_peer(), restart_session_timer(), S_OR, set_pvt_allowed_methods(), set_t38_capabilities(), sip_alreadygone(), sip_cancel_destroy(), sip_cfg, sip_methods, sip_new(), sip_pvt_lock, sip_pvt_unlock, sip_refer_allocate(), sip_scheddestroy(), sip_st_alloc(), sip_t38_abort(), ast_party_connected_line::source, st_get_mode(), st_get_refresher(), st_get_se(), start_session_timer(), ast_party_name::str, ast_party_number::str, strcasestr(), strsep(), ast_party_id::tag, cfsip_methods::text, transmit_fake_auth_response(), transmit_provisional_response(), transmit_response(), transmit_response_reliable(), transmit_response_with_minse(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), transmit_response_with_unsupported(), TRUE, unref_peer(), update_call_counter(), update_redirecting(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_incoming().

{
   int res = 1;
   int gotdest;
   const char *p_replaces;
   char *replace_id = NULL;
   int refer_locked = 0;
   const char *required;
   unsigned int required_profile = 0;
   struct ast_channel *c = NULL;    /* New channel */
   struct sip_peer *authpeer = NULL;   /* Matching Peer */
   int reinvite = 0;
   int rtn;
   struct ast_party_redirecting redirecting;
   struct ast_set_party_redirecting update_redirecting;

   const char *p_uac_se_hdr;       /* UAC's Session-Expires header string                      */
   const char *p_uac_min_se;       /* UAC's requested Min-SE interval (char string)            */
   int uac_max_se = -1;            /* UAC's Session-Expires in integer format                  */
   int uac_min_se = -1;            /* UAC's Min-SE in integer format                           */
   int st_active = FALSE;          /* Session-Timer on/off boolean                             */
   int st_interval = 0;            /* Session-Timer negotiated refresh interval                */
   enum st_refresher st_ref;       /* Session-Timer session refresher                          */
   int dlg_min_se = -1;
   struct {
      char exten[AST_MAX_EXTENSION];
      char context[AST_MAX_CONTEXT];
   } pickup = {
         .exten = "",
   };
   st_ref = SESSION_TIMER_REFRESHER_AUTO;

   /* Find out what they support */
   if (!p->sipoptions) {
      const char *supported = get_header(req, "Supported");
      if (!ast_strlen_zero(supported)) {
         p->sipoptions = parse_sip_options(supported, NULL, 0);
      }
   }

   /* Find out what they require */
   required = get_header(req, "Require");
   if (!ast_strlen_zero(required)) {
      char unsupported[256] = { 0, };
      required_profile = parse_sip_options(required, unsupported, ARRAY_LEN(unsupported));

      /* If there are any options required that we do not support,
       * then send a 420 with only those unsupported options listed */
      if (!ast_strlen_zero(unsupported)) {
         transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, unsupported);
         ast_log(LOG_WARNING, "Received SIP INVITE with unsupported required extension: required:%s unsupported:%s\n", required, unsupported);
         p->invitestate = INV_COMPLETED;
         if (!p->lastinvite)
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = -1;
         goto request_invite_cleanup;
      }
   }

   /* The option tags may be present in Supported: or Require: headers.
   Include the Require: option tags for further processing as well */
   p->sipoptions |= required_profile;
   p->reqsipoptions = required_profile;

   /* Check if this is a loop */
   if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->invitestate != INV_TERMINATED && p->invitestate != INV_CONFIRMED) && p->owner->_state != AST_STATE_UP) {
      /* This is a call to ourself.  Send ourselves an error code and stop
         processing immediately, as SIP really has no good mechanism for
         being able to call yourself */
      /* If pedantic is on, we need to check the tags. If they're different, this is
         in fact a forked call through a SIP proxy somewhere. */
      int different;
      const char *initial_rlPart2 = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
      const char *this_rlPart2 = REQ_OFFSET_TO_STR(req, rlPart2);
      if (sip_cfg.pedanticsipchecking)
         different = sip_uri_cmp(initial_rlPart2, this_rlPart2);
      else
         different = strcmp(initial_rlPart2, this_rlPart2);
      if (!different) {
         transmit_response(p, "482 Loop Detected", req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = 0;
         goto request_invite_cleanup;
      } else {
         /*! This is a spiral. What we need to do is to just change the outgoing INVITE
          * so that it now routes to the new Request URI. Since we created the INVITE ourselves
          * that should be all we need to do.
          *
          * \todo XXX This needs to be reviewed.  YOu don't change the request URI really, you route the packet
          * correctly instead...
          */
         char *uri = ast_strdupa(this_rlPart2);
         char *at = strchr(uri, '@');
         char *peerorhost;
         ast_debug(2, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", initial_rlPart2, this_rlPart2);
         transmit_response(p, "100 Trying", req);
         if (at) {
            *at = '\0';
         }
         /* Parse out "sip:" */
         if ((peerorhost = strchr(uri, ':'))) {
            *peerorhost++ = '\0';
         }
         ast_string_field_set(p, theirtag, NULL);
         /* Treat this as if there were a call forward instead...
          */
         ast_string_field_set(p->owner, call_forward, peerorhost);
         ast_queue_control(p->owner, AST_CONTROL_BUSY);
         res = 0;
         goto request_invite_cleanup;
      }
   }

   if (!req->ignore && p->pendinginvite) {
      if (!ast_test_flag(&p->flags[0], SIP_OUTGOING) && (p->invitestate == INV_COMPLETED || p->invitestate == INV_TERMINATED)) {
         /* What do these circumstances mean? We have received an INVITE for an "incoming" dialog for which we
          * have sent a final response. We have not yet received an ACK, though (which is why p->pendinginvite is non-zero).
          * We also know that the INVITE is not a retransmission, because otherwise the "ignore" flag would be set.
          * This means that either we are receiving a reinvite for a terminated dialog, or we are receiving an INVITE with
          * credentials based on one we challenged earlier.
          *
          * The action to take in either case is to treat the INVITE as though it contains an implicit ACK for the previous
          * transaction. Calling __sip_ack will take care of this by clearing the p->pendinginvite and removing the response
          * from the previous transaction from the list of outstanding packets.
          */
         __sip_ack(p, p->pendinginvite, 1, 0);
      } else {
         /* We already have a pending invite. Sorry. You are on hold. */
         p->glareinvite = seqno;
         if (p->rtp && find_sdp(req)) {
            struct ast_sockaddr addr;
            if (get_ip_and_port_from_sdp(req, SDP_AUDIO, &addr)) {
               ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.\n");
            } else {
               ast_rtp_instance_set_alt_remote_address(p->rtp, &addr);
            }
            if (p->vrtp) {
               if (get_ip_and_port_from_sdp(req, SDP_VIDEO, &addr)) {
                  ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Video may not work properly on this call.\n");
               } else {
                  ast_rtp_instance_set_alt_remote_address(p->vrtp, &addr);
               }
            }
         }
         transmit_response_reliable(p, "491 Request Pending", req);
         ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
         /* Don't destroy dialog here */
         res = 0;
         goto request_invite_cleanup;
      }
   }

   p_replaces = get_header(req, "Replaces");
   if (!ast_strlen_zero(p_replaces)) {
      /* We have a replaces header */
      char *ptr;
      char *fromtag = NULL;
      char *totag = NULL;
      char *start, *to;
      int error = 0;

      if (p->owner) {
         ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         /* Do not destroy existing call */
         res = -1;
         goto request_invite_cleanup;
      }

      if (sipdebug)
         ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces);
      /* Create a buffer we can manipulate */
      replace_id = ast_strdupa(p_replaces);
      ast_uri_decode(replace_id);

      if (!p->refer && !sip_refer_allocate(p)) {
         transmit_response_reliable(p, "500 Server Internal Error", req);
         append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         p->invitestate = INV_COMPLETED;
         res = -1;
         goto request_invite_cleanup;
      }

      /*  Todo: (When we find phones that support this)
         if the replaces header contains ";early-only"
         we can only replace the call in early
         stage, not after it's up.

         If it's not in early mode, 486 Busy.
      */

      /* Skip leading whitespace */
      replace_id = ast_skip_blanks(replace_id);

      start = replace_id;
      while ( (ptr = strsep(&start, ";")) ) {
         ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
         if ( (to = strcasestr(ptr, "to-tag=") ) )
            totag = to + 7;   /* skip the keyword */
         else if ( (to = strcasestr(ptr, "from-tag=") ) ) {
            fromtag = to + 9; /* skip the keyword */
            fromtag = strsep(&fromtag, "&"); /* trim what ? */
         }
      }

      if (sipdebug)
         ast_debug(4, "Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n",
                 replace_id,
                 fromtag ? fromtag : "<no from tag>",
                 totag ? totag : "<no to tag>");

      /* Try to find call that we are replacing.
         If we have a Replaces header, we need to cancel that call if we succeed with this call.
         First we cheat a little and look for a magic call-id from phones that support
         dialog-info+xml so we can do technology independent pickup... */
      if (strncmp(replace_id, "pickup-", 7) == 0) {
         struct sip_pvt *subscription = NULL;
         replace_id += 7; /* Worst case we are looking at \0 */

         if ((subscription = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
            ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
            transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
            error = 1;
         } else {
            ast_log(LOG_NOTICE, "Trying to pick up %s@%s\n", subscription->exten, subscription->context);
            ast_copy_string(pickup.exten, subscription->exten, sizeof(pickup.exten));
            ast_copy_string(pickup.context, subscription->context, sizeof(pickup.context));
            sip_pvt_unlock(subscription);
            if (subscription->owner) {
               ast_channel_unlock(subscription->owner);
            }
         }
      }

      /* This locks both refer_call pvt and refer_call pvt's owner!!!*/
      if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
         error = 1;
      } else {
         refer_locked = 1;
      }

      /* The matched call is the call from the transferer to Asterisk .
         We want to bridge the bridged part of the call to the
         incoming invite, thus taking over the refered call */

      if (p->refer->refer_call == p) {
         ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
         p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         error = 1;
      }

      if (!error && ast_strlen_zero(pickup.exten) && !p->refer->refer_call->owner) {
         /* Oops, someting wrong anyway, no owner, no call */
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
         /* Check for better return code */
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
         error = 1;
      }

      if (!error && ast_strlen_zero(pickup.exten) && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "603 Declined (Replaces)", req);
         error = 1;
      }

      if (error) {   /* Give up this dialog */
         append_history(p, "Xfer", "INVITE/Replace Failed.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         sip_pvt_unlock(p);
         if (p->refer->refer_call) {
            sip_pvt_unlock(p->refer->refer_call);
            if (p->refer->refer_call->owner) {
               ast_channel_unlock(p->refer->refer_call->owner);
            }
         }
         refer_locked = 0;
         p->invitestate = INV_COMPLETED;
         res = -1;
         goto request_invite_cleanup;
      }
   }

   /* Check if this is an INVITE that sets up a new dialog or
      a re-invite in an existing dialog */

   if (!req->ignore) {
      int newcall = (p->initreq.headers ? TRUE : FALSE);

      if (sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      /* This also counts as a pending invite */
      p->pendinginvite = seqno;
      check_via(p, req);

      copy_request(&p->initreq, req);     /* Save this INVITE as the transaction basis */
      if (sipdebug)
         ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      if (!p->owner) {  /* Not a re-invite */
         if (debug)
            ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
         if (newcall)
            append_history(p, "Invite", "New call: %s", p->callid);
         parse_ok_contact(p, req);
      } else { /* Re-invite on existing call */
         ast_clear_flag(&p->flags[0], SIP_OUTGOING);  /* This is now an inbound dialog */
         if (get_rpid(p, req)) {
            struct ast_party_connected_line connected;
            struct ast_set_party_connected_line update_connected;

            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));
            if (p->cid_num) {
               update_connected.id.number = 1;
               connected.id.number.valid = 1;
               connected.id.number.str = (char *) p->cid_num;
               connected.id.number.presentation = p->callingpres;
            }
            if (p->cid_name) {
               update_connected.id.name = 1;
               connected.id.name.valid = 1;
               connected.id.name.str = (char *) p->cid_name;
               connected.id.name.presentation = p->callingpres;
            }
            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         /* Handle SDP here if we already have an owner */
         if (find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_INITIATE)) {
               transmit_response_reliable(p, "488 Not acceptable here", req);
               if (!p->lastinvite)
                  sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
               res = -1;
               goto request_invite_cleanup;
            }
            ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
         } else {
            p->jointcapability = p->capability;
            ast_debug(1, "Hm....  No sdp for the moment\n");
         }
         if (p->do_history) /* This is a response, note what it was for */
            append_history(p, "ReInv", "Re-invite received");
      }
   } else if (debug)
      ast_verbose("Ignoring this INVITE request\n");

   if (!p->lastinvite && !req->ignore && !p->owner) {
      /* This is a new invite */
      /* Handle authentication if this is our first invite */
      int cc_recall_core_id = -1;
      set_pvt_allowed_methods(p, req);
      res = check_user_full(p, req, SIP_INVITE, e, XMIT_RELIABLE, addr, &authpeer);
      if (res == AUTH_CHALLENGE_SENT) {
         p->invitestate = INV_COMPLETED;     /* Needs to restart in another INVITE transaction */
         res = 0;
         goto request_invite_cleanup;
      }
      if (res < 0) { /* Something failed in authentication */
         if (res == AUTH_FAKE_AUTH) {
            ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
            transmit_fake_auth_response(p, SIP_INVITE, req, XMIT_RELIABLE);
         } else {
            ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
            transmit_response_reliable(p, "403 Forbidden", req);
         }
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         ast_string_field_set(p, theirtag, NULL);
         res = 0;
         goto request_invite_cleanup;
      }

      /* Successful authentication and peer matching so record the peer related to this pvt (for easy access to peer settings) */
      if (p->relatedpeer) {
         p->relatedpeer = unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
      }
      if (authpeer) {
         p->relatedpeer = ref_peer(authpeer, "setting dialog's relatedpeer pointer");
      }
      /* If T38 is needed but not present, then make it magically appear */
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && !p->udptl) {
         if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
            p->t38_maxdatagram = global_t38_maxdatagram;
            set_t38_capabilities(p);
         } else {
            /* udptl creation failed, T38 can not be supported on this dialog */
            ast_debug(1, "UDPTL creation failed on dialog.\n");
            ast_clear_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT);
         }
      }

      /* We have a successful authentication, process the SDP portion if there is one */
      if (find_sdp(req)) {
         if (process_sdp(p, req, SDP_T38_INITIATE)) {
            /* Unacceptable codecs */
            transmit_response_reliable(p, "488 Not acceptable here", req);
            p->invitestate = INV_COMPLETED;
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            ast_debug(1, "No compatible codecs for this SIP call.\n");
            res = -1;
            goto request_invite_cleanup;
         }
      } else { /* No SDP in invite, call control session */
         p->jointcapability = p->capability;
         ast_debug(2, "No SDP in Invite, third party call control\n");
      }

      /* Queue NULL frame to prod ast_rtp_bridge if appropriate */
      /* This seems redundant ... see !p-owner above */
      if (p->owner)
         ast_queue_frame(p->owner, &ast_null_frame);


      /* Initialize the context if it hasn't been already */
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, sip_cfg.default_context);


      /* Check number of concurrent calls -vs- incoming limit HERE */
      ast_debug(1, "Checking SIP call limits for device %s\n", p->username);
      if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
         if (res < 0) {
            ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
            transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            p->invitestate = INV_COMPLETED;
         }
         res = 0;
         goto request_invite_cleanup;
      }
      gotdest = get_destination(p, NULL, &cc_recall_core_id);  /* Get destination right away */
      extract_uri(p, req);       /* Get the Contact URI */
      build_contact(p);       /* Build our contact header */

      if (p->rtp) {
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      }

      if (!replace_id && (gotdest != SIP_GET_DEST_EXTEN_FOUND)) { /* No matching extension found */
         switch(gotdest) {
         case SIP_GET_DEST_INVALID_URI:
            transmit_response_reliable(p, "416 Unsupported URI scheme", req);
            break;
         case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
               transmit_response_reliable(p, "484 Address Incomplete", req);
               break;
            }
         /* INTENTIONAL FALL THROUGH */
         case SIP_GET_DEST_EXTEN_NOT_FOUND:
         case SIP_GET_DEST_REFUSED:
         default:
            {
               char *decoded_exten = ast_strdupa(p->exten);
               transmit_response_reliable(p, "404 Not Found", req);
               ast_uri_decode(decoded_exten);
               ast_log(LOG_NOTICE, "Call from '%s' to extension"
                  " '%s' rejected because extension not found in context '%s'.\n",
                  S_OR(p->username, p->peername), decoded_exten, p->context);
            }
         } /* end switch */

         p->invitestate = INV_COMPLETED;
         update_call_counter(p, DEC_CALL_LIMIT);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = 0;
         goto request_invite_cleanup;
      } else {

         /* If no extension was specified, use the s one */
         /* Basically for calling to IP/Host name only */
         if (ast_strlen_zero(p->exten))
            ast_string_field_set(p, exten, "s");
         /* Initialize our tag */

         make_our_tag(p->tag, sizeof(p->tag));
         /* First invitation - create the channel */
         c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL), NULL);
         if (cc_recall_core_id != -1) {
            ast_setup_cc_recall_datastore(c, cc_recall_core_id);
            ast_cc_agent_set_interfaces_chanvar(c);
         }
         *recount = 1;

         /* Save Record-Route for any later requests we make on this dialogue */
         build_route(p, req, 0);

         if (c) {
            ast_party_redirecting_init(&redirecting);
            memset(&update_redirecting, 0, sizeof(update_redirecting));
            /* Pre-lock the call */
            ast_channel_lock(c);
            change_redirecting_information(p, req, &redirecting, &update_redirecting,
               FALSE); /*Will return immediately if no Diversion header is present */
            ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
            ast_party_redirecting_free(&redirecting);
         }
      }
   } else {
      ast_party_redirecting_init(&redirecting);
      memset(&update_redirecting, 0, sizeof(update_redirecting));
      if (sipdebug) {
         if (!req->ignore)
            ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
         else
            ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
      }
      if (!req->ignore)
         reinvite = 1;
      c = p->owner;
      change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); /*Will return immediately if no Diversion header is present */
      if (c) {
         ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
      }
      ast_party_redirecting_free(&redirecting);
   }

   /* Session-Timers */
   if ((p->sipoptions & SIP_OPT_TIMER) && !ast_strlen_zero(get_header(req, "Session-Expires"))) {
      /* The UAC has requested session-timers for this session. Negotiate
      the session refresh interval and who will be the refresher */
      ast_debug(2, "Incoming INVITE with 'timer' option supported and \"Session-Expires\" header.\n");

      /* Allocate Session-Timers struct w/in the dialog */
      if (!p->stimer)
         sip_st_alloc(p);

      /* Parse the Session-Expires header */
      p_uac_se_hdr = get_header(req, "Session-Expires");
      rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
      if (rtn != 0) {
         transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
         p->invitestate = INV_COMPLETED;
         if (!p->lastinvite) {
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         }
         res = -1;
         goto request_invite_cleanup;
      }

      /* Parse the Min-SE header */
      p_uac_min_se = get_header(req, "Min-SE");
      if (!ast_strlen_zero(p_uac_min_se)) {
         rtn = parse_minse(p_uac_min_se, &uac_min_se);
         if (rtn != 0) {
            transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            res = -1;
            goto request_invite_cleanup;
         }
      }

      dlg_min_se = st_get_se(p, FALSE);
      switch (st_get_mode(p)) {
      case SESSION_TIMER_MODE_ACCEPT:
      case SESSION_TIMER_MODE_ORIGINATE:
         if (uac_max_se > 0 && uac_max_se < dlg_min_se) {
            transmit_response_with_minse(p, "422 Session Interval Too Small", req, dlg_min_se);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            res = -1;
            goto request_invite_cleanup;
         }

         p->stimer->st_active_peer_ua = TRUE;
         st_active = TRUE;
         if (st_ref == SESSION_TIMER_REFRESHER_AUTO) {
            st_ref = st_get_refresher(p);
         }

         if (uac_max_se > 0) {
            int dlg_max_se = st_get_se(p, TRUE);
            if (dlg_max_se >= uac_min_se) {
               st_interval = (uac_max_se < dlg_max_se) ? uac_max_se : dlg_max_se;
            } else {
               st_interval = uac_max_se;
            }
         } else {
            /* Set to default max value */
            st_interval = global_max_se;
         }
         break;

      case SESSION_TIMER_MODE_REFUSE:
         if (p->reqsipoptions & SIP_OPT_TIMER) {
            transmit_response_with_unsupported(p, "420 Option Disabled", req, required);
            ast_log(LOG_WARNING, "Received SIP INVITE with supported but disabled option: %s\n", required);
            p->invitestate = INV_COMPLETED;
            if (!p->lastinvite) {
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
            res = -1;
            goto request_invite_cleanup;
         }
         break;

      default:
         ast_log(LOG_ERROR, "Internal Error %d at %s:%d\n", st_get_mode(p), __FILE__, __LINE__);
         break;
      }
   } else {
      /* The UAC did not request session-timers.  Asterisk (UAS), will now decide
      (based on session-timer-mode in sip.conf) whether to run session-timers for
      this session or not. */
      switch (st_get_mode(p)) {
      case SESSION_TIMER_MODE_ORIGINATE:
         st_active = TRUE;
         st_interval = st_get_se(p, TRUE);
         st_ref = SESSION_TIMER_REFRESHER_UAS;
         p->stimer->st_active_peer_ua = FALSE;
         break;

      default:
         break;
      }
   }

   if (reinvite == 0) {
      /* Session-Timers: Start session refresh timer based on negotiation/config */
      if (st_active == TRUE) {
         p->stimer->st_active   = TRUE;
         p->stimer->st_interval = st_interval;
         p->stimer->st_ref      = st_ref;
         start_session_timer(p);
      }
   } else {
      if (p->stimer->st_active == TRUE) {
         /* Session-Timers:  A re-invite request sent within a dialog will serve as
         a refresh request, no matter whether the re-invite was sent for refreshing
         the session or modifying it.*/
         ast_debug (2, "Restarting session-timers on a refresh - %s\n", p->callid);

         /* The UAC may be adjusting the session-timers mid-session */
         if (st_interval > 0) {
            p->stimer->st_interval = st_interval;
            p->stimer->st_ref      = st_ref;
         }

         restart_session_timer(p);
         if (p->stimer->st_expirys > 0) {
            p->stimer->st_expirys--;
         }
      }
   }

   if (!req->ignore && p)
      p->lastinvite = seqno;

   if (replace_id) { /* Attended transfer or call pickup - we're the target */
      if (!ast_strlen_zero(pickup.exten)) {
         append_history(p, "Xfer", "INVITE/Replace received");

         /* Let the caller know we're giving it a shot */
         transmit_response(p, "100 Trying", req);
         p->invitestate = INV_PROCEEDING;
         ast_setstate(c, AST_STATE_RING);

         /* Do the pickup itself */
         ast_channel_unlock(c);
         *nounlock = 1;

         /* since p->owner (c) is unlocked, we need to go ahead and unlock pvt for both
          * magic pickup and ast_hangup.  Both of these functions will attempt to lock
          * p->owner again, which can cause a deadlock if we already hold a lock on p.
          * Locking order is, channel then pvt.  Dead lock avoidance must be used if
          * called the other way around. */
         sip_pvt_unlock(p);
         do_magic_pickup(c, pickup.exten, pickup.context);
         /* Now we're either masqueraded or we failed to pickup, in either case we... */
         ast_hangup(c);
         sip_pvt_lock(p); /* pvt is expected to remain locked on return, so re-lock it */

         res = 0;
         goto request_invite_cleanup;
      } else {
         /* Go and take over the target call */
         if (sipdebug)
            ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
         res = handle_invite_replaces(p, req, debug, seqno, addr, nounlock);
         refer_locked = 0;
         goto request_invite_cleanup;
      }
   }


   if (c) { /* We have a call  -either a new call or an old one (RE-INVITE) */
      enum ast_channel_state c_state = c->_state;

      if (c_state != AST_STATE_UP && reinvite &&
         (p->invitestate == INV_TERMINATED || p->invitestate == INV_CONFIRMED)) {
         /* If these conditions are true, and the channel is still in the 'ringing'
          * state, then this likely means that we have a situation where the initial
          * INVITE transaction has completed *but* the channel's state has not yet been
          * changed to UP. The reason this could happen is if the reinvite is received
          * on the SIP socket prior to an application calling ast_read on this channel
          * to read the answer frame we earlier queued on it. In this case, the reinvite
          * is completely legitimate so we need to handle this the same as if the channel
          * were already UP. Thus we are purposely falling through to the AST_STATE_UP case.
          */
         c_state = AST_STATE_UP;
      }

      switch(c_state) {
      case AST_STATE_DOWN:
         ast_debug(2, "%s: New call is still down.... Trying... \n", c->name);
         transmit_provisional_response(p, "100 Trying", req, 0);
         p->invitestate = INV_PROCEEDING;
         ast_setstate(c, AST_STATE_RING);
         if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
            enum ast_pbx_result result;

            result = ast_pbx_start(c);

            switch(result) {
            case AST_PBX_FAILED:
               ast_log(LOG_WARNING, "Failed to start PBX :(\n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "503 Unavailable", req);
               break;
            case AST_PBX_CALL_LIMIT:
               ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "480 Temporarily Unavailable", req);
               break;
            case AST_PBX_SUCCESS:
               /* nothing to do */
               break;
            }

            if (result) {

               /* Unlock locks so ast_hangup can do its magic */
               ast_channel_unlock(c);
               sip_pvt_unlock(p);
               ast_hangup(c);
               sip_pvt_lock(p);
               c = NULL;
            }
         } else { /* Pickup call in call group */
            ast_channel_unlock(c);
            *nounlock = 1;
            if (ast_pickup_call(c)) {
               ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
               transmit_response_reliable(p, "503 Unavailable", req);
               sip_alreadygone(p);
               /* Unlock locks so ast_hangup can do its magic */
               sip_pvt_unlock(p);
               c->hangupcause = AST_CAUSE_CALL_REJECTED;
            } else {
               sip_pvt_unlock(p);
               c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
            }
            p->invitestate = INV_COMPLETED;
            ast_hangup(c);
            sip_pvt_lock(p);
            c = NULL;
         }
         break;
      case AST_STATE_RING:
         transmit_provisional_response(p, "100 Trying", req, 0);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_RINGING:
         transmit_provisional_response(p, "180 Ringing", req, 0);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_UP:
         ast_debug(2, "%s: This call is UP.... \n", c->name);

         transmit_response(p, "100 Trying", req);

         if (p->t38.state == T38_PEER_REINVITE) {
            p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, dialog_ref(p, "passing dialog ptr into sched structure based on t38id for sip_t38_abort."));
         } else if (p->t38.state == T38_ENABLED) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)));
         } else if (p->t38.state == T38_DISABLED) {
            /* If this is not a re-invite or something to ignore - it's critical */
            if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
               ast_log(LOG_WARNING, "Target does not support required crypto\n");
               transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
            } else {
               ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
               transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
            }
         }

         p->invitestate = INV_TERMINATED;
         break;
      default:
         ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
         transmit_response(p, "100 Trying", req);
         break;
      }
   } else {
      if (p && (p->autokillid == -1)) {
         const char *msg;

         if (!p->jointcapability)
            msg = "488 Not Acceptable Here (codec error)";
         else {
            ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
            msg = "503 Unavailable";
         }
         transmit_response_reliable(p, msg, req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
   }

request_invite_cleanup:

   if (refer_locked && p->refer && p->refer->refer_call) {
      sip_pvt_unlock(p->refer->refer_call);
      if (p->refer->refer_call->owner) {
         ast_channel_unlock(p->refer->refer_call->owner);
      }
   }
   if (authpeer) {
      authpeer = unref_peer(authpeer, "unref_peer, from handle_request_invite authpeer");
   }

   return res;
}

static int handle_request_message	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Handle incoming MESSAGE request.

Definition at line 22382 of file chan_sip.c.

References ast_verbose, receive_message(), and transmit_response().

Referenced by handle_incoming().

{
   if (!req->ignore) {
      if (req->debug)
         ast_verbose("Receiving message!\n");
      receive_message(p, req);
   } else
      transmit_response(p, "202 Accepted", req);
   return 1;
}

static int handle_request_notify	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		int	seqno,
		const char *	e
	)			[static]

Handle incoming notifications.

Definition at line 20329 of file chan_sip.c.

References AST_CONTROL_TRANSFER, ast_debug, AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_IE_PLTYPE_UINT, AST_EVENT_MWI, ast_event_new(), ast_event_queue_and_cache(), ast_log(), ast_queue_control_data(), ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), AST_TRANSFER_FAILED, AST_TRANSFER_SUCCESS, FALSE, find_peer(), get_body(), get_header(), get_msg_text(), handle_cc_notify(), LOG_NOTICE, LOG_WARNING, mailbox, sip_scheddestroy(), strsep(), transmit_response(), TRUE, and unref_peer().

Referenced by handle_incoming().

{
   /* This is mostly a skeleton for future improvements */
   /* Mostly created to return proper answers on notifications on outbound REFER's */
   int res = 0;
   const char *event = get_header(req, "Event");
   char *eventid = NULL;
   char *sep;

   if( (sep = strchr(event, ';')) ) {  /* XXX bug here - overwriting string ? */
      *sep++ = '\0';
      eventid = sep;
   }
   
   if (sipdebug)
      ast_debug(2, "Got NOTIFY Event: %s\n", event);

   if (!strcmp(event, "refer")) {
      /* Save nesting depth for now, since there might be other events we will
         support in the future */

      /* Handle REFER notifications */

      char buf[1024];
      char *cmd, *code;
      int respcode;
      int success = TRUE;

      /* EventID for each transfer... EventID is basically the REFER cseq

       We are getting notifications on a call that we transfered
       We should hangup when we are getting a 200 OK in a sipfrag
       Check if we have an owner of this event */
      
      /* Check the content type */
      if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
         /* We need a sipfrag */
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /* Get the text of the attachment */
      if (get_msg_text(buf, sizeof(buf), req, TRUE)) {
         ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /*
      From the RFC...
      A minimal, but complete, implementation can respond with a single
      NOTIFY containing either the body:
         SIP/2.0 100 Trying
      
      if the subscription is pending, the body:
         SIP/2.0 200 OK
      if the reference was successful, the body:
         SIP/2.0 503 Service Unavailable
      if the reference failed, or the body:
         SIP/2.0 603 Declined

      if the REFER request was accepted before approval to follow the
      reference could be obtained and that approval was subsequently denied
      (see Section 2.4.7).
      
      If there are several REFERs in the same dialog, we need to
      match the ID of the event header...
      */
      ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf);
      cmd = ast_skip_blanks(buf);
      code = cmd;
      /* We are at SIP/2.0 */
      while(*code && (*code > 32)) {   /* Search white space */
         code++;
      }
      *code++ = '\0';
      code = ast_skip_blanks(code);
      sep = code;
      sep++;
      while(*sep && (*sep > 32)) {  /* Search white space */
         sep++;
      }
      *sep++ = '\0';       /* Response string */
      respcode = atoi(code);
      switch (respcode) {
      case 200:   /* OK: The new call is up, hangup this call */
         /* Hangup the call that we are replacing */
         break;
      case 301: /* Moved permenantly */
      case 302: /* Moved temporarily */
         /* Do we get the header in the packet in this case? */
         success = FALSE;
         break;
      case 503:   /* Service Unavailable: The new call failed */
      case 603:   /* Declined: Not accepted */
            /* Cancel transfer, continue the current call */
         success = FALSE;
         break;
      case 0:     /* Parse error */
            /* Cancel transfer, continue the current call */
         ast_log(LOG_NOTICE, "Error parsing sipfrag in NOTIFY in response to REFER.\n");
         success = FALSE;
         break;
      default:
         if (respcode < 200) {
            /* ignore provisional responses */
            success = -1;
         } else {
            ast_log(LOG_NOTICE, "Got unknown code '%d' in NOTIFY in response to REFER.\n", respcode);
            success = FALSE;
         }
         break;
      }
      if (success == FALSE) {
         ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n");
      }

      if (p->owner && success != -1) {
         enum ast_control_transfer message = success ? AST_TRANSFER_SUCCESS : AST_TRANSFER_FAILED;
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      /* Confirm that we received this packet */
      transmit_response(p, "200 OK", req);
   } else if (!strcmp(event, "message-summary")) {
      const char *mailbox = NULL;
      char *c = ast_strdupa(get_body(req, "Voice-Message", ':'));

      if (!p->mwi) {
         struct sip_peer *peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);

         if (peer) {
            mailbox = ast_strdupa(peer->unsolicited_mailbox);
            unref_peer(peer, "removing unsolicited mwi ref");
         }
      } else {
         mailbox = p->mwi->mailbox;
      }

      if (!ast_strlen_zero(mailbox) && !ast_strlen_zero(c)) {
         char *old = strsep(&c, " ");
         char *new = strsep(&old, "/");
         struct ast_event *event;

         if ((event = ast_event_new(AST_EVENT_MWI,
                     AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox,
                     AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, "SIP_Remote",
                     AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(new),
                     AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(old),
                     AST_EVENT_IE_END))) {
            ast_event_queue_and_cache(event);
         }
         transmit_response(p, "200 OK", req);
      } else {
         transmit_response(p, "489 Bad event", req);
         res = -1;
      }
   } else if (!strcmp(event, "keep-alive")) {
       /* Used by Sipura/Linksys for NAT pinhole,
        * just confirm that we recieved the packet. */
      transmit_response(p, "200 OK", req);
   } else if (!strcmp(event, "call-completion")) {
      res = handle_cc_notify(p, req);
   } else {
      /* We don't understand this event. */
      transmit_response(p, "489 Bad event", req);
      res = -1;
   }

   if (!p->lastinvite)
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return res;
}

static int handle_request_options	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const char *	e
	)			[static]

Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.

Definition at line 20508 of file chan_sip.c.

References ast_log(), ast_shutting_down(), ast_string_field_set, ast_strlen_zero(), build_contact(), check_user(), context, copy_request(), get_destination(), get_header(), LOG_NOTICE, set_pvt_allowed_methods(), sip_cfg, sip_scheddestroy(), transmit_fake_auth_response(), transmit_response(), and transmit_response_with_allow().

Referenced by handle_incoming().

{
   int res;

   if (p->lastinvite) {
      /* if this is a request in an active dialog, just confirm that the dialog exists. */
      transmit_response_with_allow(p, "200 OK", req, 0);
      return 0;
   }

   if (sip_cfg.auth_options_requests) {
      /* Do authentication if this OPTIONS request began the dialog */
      copy_request(&p->initreq, req);
      set_pvt_allowed_methods(p, req);
      res = check_user(p, req, SIP_OPTIONS, e, XMIT_UNRELIABLE, addr);
      if (res == AUTH_CHALLENGE_SENT) {
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      }
      if (res < 0) { /* Something failed in authentication */
         if (res == AUTH_FAKE_AUTH) {
            ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
            transmit_fake_auth_response(p, SIP_OPTIONS, req, XMIT_UNRELIABLE);
         } else {
            ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
            transmit_response(p, "403 Forbidden", req);
         }
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      }
   }

   /* must go through authentication before getting here */
   res = (get_destination(p, req, NULL) == SIP_GET_DEST_EXTEN_FOUND ? 0 : -1);
   build_contact(p);

   if (ast_strlen_zero(p->context))
      ast_string_field_set(p, context, sip_cfg.default_context);

   if (ast_shutting_down())
      transmit_response_with_allow(p, "503 Unavailable", req, 0);
   else if (res < 0)
      transmit_response_with_allow(p, "404 Not Found", req, 0);
   else
      transmit_response_with_allow(p, "200 OK", req, 0);

   /* Destroy if this OPTIONS was the opening request, but not if
      it's in the middle of a normal call flow. */
   sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return res;
}

static int handle_request_publish	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const int	seqno,
		const char *	uri
	)			[static]

Definition at line 22813 of file chan_sip.c.

References __sip_ack(), ast_log(), ast_string_field_set, ast_strlen_zero(), check_user(), determine_sip_publish_type(), get_esc(), get_header(), handle_sip_publish_initial(), handle_sip_publish_modify(), handle_sip_publish_refresh(), handle_sip_publish_remove(), LOG_NOTICE, sip_scheddestroy(), transmit_fake_auth_response(), transmit_response(), and transmit_response_reliable().

Referenced by handle_incoming().

{
   const char *etag = get_header(req, "SIP-If-Match");
   const char *event = get_header(req, "Event");
   struct event_state_compositor *esc;
   enum sip_publish_type publish_type;
   const char *expires_str = get_header(req, "Expires");
   int expires_int;
   int auth_result;
   int handler_result = -1;

   if (ast_strlen_zero(event)) {
      transmit_response(p, "489 Bad Event", req);
      return -1;
   }

   if (!(esc = get_esc(event))) {
      transmit_response(p, "489 Bad Event", req);
      return -1;
   }

   auth_result = check_user(p, req, SIP_PUBLISH, uri, XMIT_RELIABLE, addr);
   if (auth_result == AUTH_CHALLENGE_SENT) {
      p->lastinvite = seqno;
      return 0;
   } else if (auth_result < 0) {
      if (auth_result == AUTH_FAKE_AUTH) {
         ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
         transmit_fake_auth_response(p, SIP_INVITE, req, XMIT_RELIABLE);
      } else {
         ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
         transmit_response_reliable(p, "403 Forbidden", req);
      }
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_string_field_set(p, theirtag, NULL);
      return 0;
   } else if (auth_result == AUTH_SUCCESSFUL && p->lastinvite) {
      /* We need to stop retransmitting the 401 */
      __sip_ack(p, p->lastinvite, 1, 0);
   }

   publish_type = determine_sip_publish_type(req, event, etag, expires_str, &expires_int);

   /* It is the responsibility of these handlers to formulate any response
    * sent for a PUBLISH
    */
   switch (publish_type) {
   case SIP_PUBLISH_UNKNOWN:
      transmit_response(p, "400 Bad Request", req);
      break;
   case SIP_PUBLISH_INITIAL:
      handler_result = handle_sip_publish_initial(p, req, esc, expires_int);
      break;
   case SIP_PUBLISH_REFRESH:
      handler_result = handle_sip_publish_refresh(p, req, esc, etag, expires_int);
      break;
   case SIP_PUBLISH_MODIFY:
      handler_result = handle_sip_publish_modify(p, req, esc, etag, expires_int);
      break;
   case SIP_PUBLISH_REMOVE:
      handler_result = handle_sip_publish_remove(p, req, esc, etag);
      break;
   default:
      transmit_response(p, "400 Impossible Condition", req);
      break;
   }

   return handler_result;
}

static int handle_request_refer	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	debug,
		int	seqno,
		int *	nounlock
	)			[static]

Definition at line 21881 of file chan_sip.c.

References append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_NORMAL_CLEARING, AST_CEL_ATTENDEDTRANSFER, AST_CEL_BLINDTRANSFER, ast_cel_report_event(), ast_channel_ref, ast_channel_trylock, ast_channel_unlock, ast_channel_unref, ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, AST_LIST_EMPTY, ast_manager_event_multichan, ast_parking_ext(), ast_queue_control(), ast_set_flag, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose, check_sip_domain(), context, copy_request(), EVENT_FLAG_CALL, FALSE, get_refer_info(), local_attended_transfer(), ast_channel::name, pbx_builtin_setvar_helper(), pvt_set_needdestroy(), sip_alreadygone(), sip_cfg, sip_park(), sip_pvt_lock, sip_pvt_unlock, sip_refer_allocate(), transmit_notify_with_sipfrag(), transmit_response(), and TRUE.

Referenced by handle_incoming().

{
   struct sip_dual current;   /* Chan1: Call between asterisk and transferer */
               /* Chan2: Call between asterisk and transferee */

   int res = 0;
   struct ast_channel *chans[2];
   current.req.data = NULL;

   if (req->debug)
      ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");

   if (!p->owner) {
      /* This is a REFER outside of an existing SIP dialog */
      /* We can't handle that, so decline it */
      ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
      transmit_response(p, "603 Declined (No dialog)", req);
      if (!req->ignore) {
         append_history(p, "Xfer", "Refer failed. Outside of dialog.");
         sip_alreadygone(p);
         pvt_set_needdestroy(p, "outside of dialog");
      }
      return 0;
   }


   /* Check if transfer is allowed from this device */
   if (p->allowtransfer == TRANSFER_CLOSED ) {
      /* Transfer not allowed, decline */
      transmit_response(p, "603 Declined (policy)", req);
      append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
      /* Do not destroy SIP session */
      return 0;
   }

   if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
      /* Already have a pending REFER */
      transmit_response(p, "491 Request pending", req);
      append_history(p, "Xfer", "Refer failed. Request pending.");
      return 0;
   }

   /* Allocate memory for call transfer data */
   if (!p->refer && !sip_refer_allocate(p)) {
      transmit_response(p, "500 Internal Server Error", req);
      append_history(p, "Xfer", "Refer failed. Memory allocation error.");
      return -3;
   }

   res = get_refer_info(p, req); /* Extract headers */

   p->refer->status = REFER_SENT;

   if (res != 0) {
      switch (res) {
      case -2: /* Syntax error */
         transmit_response(p, "400 Bad Request (Refer-to missing)", req);
         append_history(p, "Xfer", "Refer failed. Refer-to missing.");
         if (req->debug)
            ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n");
         break;
      case -3:
         transmit_response(p, "603 Declined (Non sip: uri)", req);
         append_history(p, "Xfer", "Refer failed. Non SIP uri");
         if (req->debug)
            ast_debug(1, "SIP transfer to non-SIP uri denied\n");
         break;
      default:
         /* Refer-to extension not found, fake a failed transfer */
         transmit_response(p, "202 Accepted", req);
         append_history(p, "Xfer", "Refer failed. Bad extension.");
         transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
         ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
         if (req->debug)
            ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
         break;
      }
      return 0;
   }
   if (ast_strlen_zero(p->context))
      ast_string_field_set(p, context, sip_cfg.default_context);

   /* If we do not support SIP domains, all transfers are local */
   if (sip_cfg.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      p->refer->localtransfer = 1;
      if (sipdebug)
         ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
   } else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      /* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */
      p->refer->localtransfer = 1;
   } else if (sipdebug)
         ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);

   /* Is this a repeat of a current request? Ignore it */
   /* Don't know what else to do right now. */
   if (req->ignore)
      return res;

   /* If this is a blind transfer, we have the following
   channels to work with:
   - chan1, chan2: The current call between transferer and transferee (2 channels)
   - target_channel: A new call from the transferee to the target (1 channel)
   We need to stay tuned to what happens in order to be able
   to bring back the call to the transferer */

   /* If this is a attended transfer, we should have all call legs within reach:
   - chan1, chan2: The call between the transferer and transferee (2 channels)
   - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels)
   We want to bridge chan2 with targetcall_pvt!
   
   The replaces call id in the refer message points
   to the call leg between Asterisk and the transferer.
   So we need to connect the target and the transferee channel
   and hangup the two other channels silently
   
   If the target is non-local, the call ID could be on a remote
   machine and we need to send an INVITE with replaces to the
   target. We basically handle this as a blind transfer
   and let the sip_call function catch that we need replaces
   header in the INVITE.
   */


   /* Get the transferer's channel */
   chans[0] = current.chan1 = p->owner;

   /* Find the other part of the bridge (2) - transferee */
   chans[1] = current.chan2 = ast_bridged_channel(current.chan1);

   if (sipdebug)
      ast_debug(3, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>");

   if (!current.chan2 && !p->refer->attendedtransfer) {
      /* No bridged channel, propably IVR or echo or similar... */
      /* Guess we should masquerade or something here */
      /* Until we figure it out, refuse transfer of such calls */
      if (sipdebug)
         ast_debug(3, "Refused SIP transfer on non-bridged channel.\n");
      p->refer->status = REFER_FAILED;
      append_history(p, "Xfer", "Refer failed. Non-bridged channel.");
      transmit_response(p, "603 Declined", req);
      return -1;
   }

   if (current.chan2) {
      if (sipdebug)
         ast_debug(4, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name);

      ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
   }

   ast_set_flag(&p->flags[0], SIP_GOTREFER);

   /* Attended transfer: Find all call legs and bridge transferee with target*/
   if (p->refer->attendedtransfer) {
      if ((res = local_attended_transfer(p, &current, req, seqno, nounlock)))
         return res; /* We're done with the transfer */
      /* Fall through for remote transfers that we did not find locally */
      if (sipdebug)
         ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n");
      /* Fallthrough if we can't find the call leg internally */
   }


   /* Parking a call */
   if (p->refer->localtransfer && !strcmp(p->refer->refer_to, ast_parking_ext())) {
      /* Must release c's lock now, because it will not longer be accessible after the transfer! */
      *nounlock = 1;
      ast_channel_unlock(current.chan1);
      copy_request(&current.req, req);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);
      p->refer->status = REFER_200OK;
      append_history(p, "Xfer", "REFER to call parking.");
      ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans, "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransfer2Parking: Yes\r\n",
         current.chan1->name,
         current.chan1->uniqueid,
         p->callid,
         current.chan2->name,
         current.chan2->uniqueid,
         p->refer->refer_to);
      if (sipdebug)
         ast_debug(4, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name);
      sip_park(current.chan2, current.chan1, req, seqno);
      return res;
   }

   /* Blind transfers and remote attended xfers */
   transmit_response(p, "202 Accepted", req);

   if (current.chan1 && current.chan2) {
      ast_debug(3, "chan1->name: %s\n", current.chan1->name);
      pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name);
   }
   if (current.chan2) {
      pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name);
      pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain);
      pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes");
      /* One for the new channel */
      pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes");
      /* Attended transfer to remote host, prepare headers for the INVITE */
      if (p->refer->referred_by)
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by);
   }
   /* Generate a Replaces string to be used in the INVITE during attended transfer */
   if (!ast_strlen_zero(p->refer->replaces_callid)) {
      char tempheader[SIPBUFSIZE];
      snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid,
            p->refer->replaces_callid_totag ? ";to-tag=" : "",
            p->refer->replaces_callid_totag,
            p->refer->replaces_callid_fromtag ? ";from-tag=" : "",
            p->refer->replaces_callid_fromtag);
      if (current.chan2)
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader);
   }
   /* Must release lock now, because it will not longer
      be accessible after the transfer! */
   *nounlock = 1;
   /*
    * Increase ref count so that we can delay channel destruction until after
    * we get a chance to fire off some events.
    */
   ast_channel_ref(current.chan1);
   ast_channel_unlock(current.chan1);

   /* Connect the call */

   /* FAKE ringing if not attended transfer */
   if (!p->refer->attendedtransfer)
      transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE);

   /* For blind transfer, this will lead to a new call */
   /* For attended transfer to remote host, this will lead to
      a new SIP call with a replaces header, if the dial plan allows it
   */
   if (!current.chan2) {
      /* We have no bridge, so we're talking with Asterisk somehow */
      /* We need to masquerade this call */
      /* What to do to fix this situation:
         * Set up the new call in a new channel
         * Let the new channel masq into this channel
         Please add that code here :-)
      */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      append_history(p, "Xfer", "Refer failed (only bridged calls).");
      ast_channel_unref(current.chan1);
      return -1;
   }
   ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */


   /* For blind transfers, move the call to the new extensions. For attended transfers on multiple
      servers - generate an INVITE with Replaces. Either way, let the dial plan decided  */
   res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);

   if (!res) {
      ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans, "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransferContext: %s\r\n",
         current.chan1->name,
         current.chan1->uniqueid,
         p->callid,
         current.chan2->name,
         current.chan2->uniqueid,
         p->refer->refer_to, p->refer->refer_to_context);
      /* Success  - we have a new channel */
      ast_debug(3, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind");

      while (ast_channel_trylock(current.chan1)) {
         sip_pvt_unlock(p);
         sched_yield();
         sip_pvt_lock(p);
      }

      /* XXX - what to we put in CEL 'extra' for attended transfers to external systems? NULL for now */
      ast_cel_report_event(current.chan1, p->refer->attendedtransfer? AST_CEL_ATTENDEDTRANSFER : AST_CEL_BLINDTRANSFER, NULL, p->refer->attendedtransfer ? NULL : p->refer->refer_to, current.chan2);
      ast_channel_unlock(current.chan1);

      transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE);
      if (p->refer->localtransfer)
         p->refer->status = REFER_200OK;
      if (p->owner)
         p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING;
      append_history(p, "Xfer", "Refer succeeded.");
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      /* Do not hangup call, the other side do that when we say 200 OK */
      /* We could possibly implement a timer here, auto congestion */
      res = 0;
   } else {
      ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */
      ast_debug(3, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind");
      append_history(p, "Xfer", "Refer failed.");
      /* Failure of some kind */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      res = -1;
   }

   ast_channel_unref(current.chan1);

   return res;
}

static int handle_request_register	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	sin,
		const char *	e
	)			[static]

Handle incoming REGISTER request.

Definition at line 23336 of file chan_sip.c.

References append_history, ast_debug, ast_log(), ast_sockaddr_stringify(), check_via(), copy_request(), get_header(), LOG_NOTICE, register_verify(), sip_methods, sip_scheddestroy(), and cfsip_methods::text.

Referenced by handle_incoming().

{
   enum check_auth_result res;

   /* Use this as the basis */
   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);
   if ((res = register_verify(p, addr, req, e)) < 0) {
      const char *reason;

      switch (res) {
      case AUTH_SECRET_FAILED:
         reason = "Wrong password";
         break;
      case AUTH_USERNAME_MISMATCH:
         reason = "Username/auth name mismatch";
         break;
      case AUTH_NOT_FOUND:
         reason = "No matching peer found";
         break;
      case AUTH_UNKNOWN_DOMAIN:
         reason = "Not a local domain";
         break;
      case AUTH_PEER_NOT_DYNAMIC:
         reason = "Peer is not supposed to register";
         break;
      case AUTH_ACL_FAILED:
         reason = "Device does not match ACL";
         break;
      case AUTH_BAD_TRANSPORT:
         reason = "Device not configured to use this transport type";
         break;
      default:
         reason = "Unknown failure";
         break;
      }
      ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n",
         get_header(req, "To"), ast_sockaddr_stringify(addr),
         reason);
      append_history(p, "RegRequest", "Failed : Account %s : %s", get_header(req, "To"), reason);
   } else
      append_history(p, "RegRequest", "Succeeded : Account %s", get_header(req, "To"));

   if (res < 1) {
      /* Destroy the session, but keep us around for just a bit in case they don't
         get our 200 OK */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }
   return res;
}

static int handle_request_subscribe	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		int	seqno,
		const char *	e
	)			[static]

Handle incoming SUBSCRIBE request.

Definition at line 22953 of file chan_sip.c.

References __get_header(), add_peer_mwi_subs(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_t_ref, ao2_unlock, append_history, ast_debug, AST_EXTENSION_REMOVED, ast_extension_state(), ast_extension_state2str(), ast_extension_state_add(), ast_extension_state_del(), AST_LIST_EMPTY, ast_log(), ast_set_flag, ast_sockaddr_stringify(), ast_strdupa, ast_string_field_build, ast_strlen_zero(), ast_test_flag, ast_verbose, build_contact(), build_route(), cb_extensionstate(), check_user_full(), check_via(), copy_request(), dialog_unlink_all(), FALSE, get_destination(), get_header(), gettag(), handle_cc_subscribe(), LOG_DEBUG, LOG_NOTICE, LOG_WARNING, make_our_tag(), NONE, option_debug, parse_ok_contact(), pvt_set_needdestroy(), ref_peer(), set_pvt_allowed_methods(), sip_cancel_destroy(), sip_cfg, sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), sip_send_mwi_to_peer(), cfsip_methods::text, transmit_fake_auth_response(), transmit_response(), transmit_response_reliable(), transmit_response_with_minexpires(), transmit_state_notify(), TRUE, and unref_peer().

Referenced by handle_incoming().

{
   int gotdest = 0;
   int res = 0;
   int firststate = AST_EXTENSION_REMOVED;
   struct sip_peer *authpeer = NULL;
   const char *eventheader = get_header(req, "Event");   /* Get Event package name */
   int resubscribe = (p->subscribed != NONE);
   char *temp, *event;
   struct ao2_iterator i;

   if (p->initreq.headers) {  
      /* We already have a dialog */
      if (p->initreq.method != SIP_SUBSCRIBE) {
         /* This is a SUBSCRIBE within another SIP dialog, which we do not support */
         /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
         transmit_response(p, "403 Forbidden (within dialog)", req);
         /* Do not destroy session, since we will break the call if we do */
         ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
         return 0;
      } else if (req->debug) {
         if (resubscribe)
            ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid);
         else
            ast_debug(1, "Got a new subscription %s (possibly with auth)\n", p->callid);
      }
   }

   /* Check if we have a global disallow setting on subscriptions.
      if so, we don't have to check peer settings after auth, which saves a lot of processing
   */
   if (!sip_cfg.allowsubscribe) {
      transmit_response(p, "403 Forbidden (policy)", req);
      pvt_set_needdestroy(p, "forbidden");
      return 0;
   }

   if (!req->ignore && !resubscribe) { /* Set up dialog, new subscription */
      const char *to = get_header(req, "To");
      char totag[128];
      set_pvt_allowed_methods(p, req);

      /* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */
      if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) {
         if (req->debug)
            ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n");
         transmit_response(p, "481 Subscription does not exist", req);
         pvt_set_needdestroy(p, "subscription does not exist");
         return 0;
      }

      /* Use this as the basis */
      if (req->debug)
         ast_verbose("Creating new subscription\n");

      copy_request(&p->initreq, req);
      if (sipdebug)
         ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      check_via(p, req);
      build_route(p, req, 0);
   } else if (req->debug && req->ignore)
      ast_verbose("Ignoring this SUBSCRIBE request\n");

   /* Find parameters to Event: header value and remove them for now */
   if (ast_strlen_zero(eventheader)) {
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: <none>\n");
      pvt_set_needdestroy(p, "unknown event package in subscribe");
      return 0;
   }

   if ( (strchr(eventheader, ';'))) {
      event = ast_strdupa(eventheader);   /* Since eventheader is a const, we can't change it */
      temp = strchr(event, ';');       
      *temp = '\0';           /* Remove any options for now */
                     /* We might need to use them later :-) */
   } else
      event = (char *) eventheader;    /* XXX is this legal ? */

   /* Handle authentication */
   res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, addr, &authpeer);
   /* if an authentication response was sent, we are done here */
   if (res == AUTH_CHALLENGE_SENT)  /* authpeer = NULL here */
      return 0;
   if (res < 0) {
      if (res == AUTH_FAKE_AUTH) {
         ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
         transmit_fake_auth_response(p, SIP_SUBSCRIBE, req, XMIT_UNRELIABLE);
      } else {
         ast_log(LOG_NOTICE, "Failed to authenticate device %s for SUBSCRIBE\n", get_header(req, "From"));
         transmit_response_reliable(p, "403 Forbidden", req);
      }
      pvt_set_needdestroy(p, "authentication failed");
      return 0;
   }

   /* At this point, authpeer cannot be NULL. Remember we hold a reference,
    * so we must release it when done.
    * XXX must remove all the checks for authpeer == NULL.
    */

   /* Check if this device  is allowed to subscribe at all */
   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
      transmit_response(p, "403 Forbidden (policy)", req);
      pvt_set_needdestroy(p, "subscription not allowed");
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 1)");
      return 0;
   }

   if (strcmp(event, "message-summary") && strcmp(event, "call-completion")) {
      /* Get destination right away */
      gotdest = get_destination(p, NULL, NULL);
   }

   /* Get full contact header - this needs to be used as a request URI in NOTIFY's */
   parse_ok_contact(p, req);

   build_contact(p);
   if (gotdest != SIP_GET_DEST_EXTEN_FOUND) {
      if (gotdest == SIP_GET_DEST_INVALID_URI) {
         transmit_response(p, "416 Unsupported URI scheme", req);
      } else {
         transmit_response(p, "404 Not Found", req);
      }
      pvt_set_needdestroy(p, "subscription target not found");
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");
      return 0;
   }

   /* Initialize tag for new subscriptions */   
   if (ast_strlen_zero(p->tag))
      make_our_tag(p->tag, sizeof(p->tag));

   if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
      unsigned int pidf_xml;
      const char *accept;
      int start = 0;
      enum subscriptiontype subscribed = NONE;
      const char *unknown_acceptheader = NULL;

      if (authpeer)  /* We do not need the authpeer any more */
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");

      /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
      accept = __get_header(req, "Accept", &start);
      while ((subscribed == NONE) && !ast_strlen_zero(accept)) {
         pidf_xml = strstr(accept, "application/pidf+xml") ? 1 : 0;

         /* Older versions of Polycom firmware will claim pidf+xml, but really
          * they only support xpidf+xml. */
         if (pidf_xml && strstr(p->useragent, "Polycom")) {
            subscribed = XPIDF_XML;
         } else if (pidf_xml) {
            subscribed = PIDF_XML;         /* RFC 3863 format */
         } else if (strstr(accept, "application/dialog-info+xml")) {
            subscribed = DIALOG_INFO_XML;
            /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
         } else if (strstr(accept, "application/cpim-pidf+xml")) {
            subscribed = CPIM_PIDF_XML;    /* RFC 3863 format */
         } else if (strstr(accept, "application/xpidf+xml")) {
            subscribed = XPIDF_XML;        /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
         } else {
            unknown_acceptheader = accept;
         }
         /* check to see if there is another Accept header present */
         accept = __get_header(req, "Accept", &start);
      }

      if (!start) {
         if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */
            transmit_response(p, "489 Bad Event", req);
            ast_log(LOG_WARNING,"SUBSCRIBE failure: no Accept header: pvt: "
               "stateid: %d, laststate: %d, dialogver: %d, subscribecont: "
               "'%s', subscribeuri: '%s'\n",
               p->stateid,
               p->laststate,
               p->dialogver,
               p->subscribecontext,
               p->subscribeuri);
            pvt_set_needdestroy(p, "no Accept header");
            return 0;
         }
         /* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least.
            so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */
      } else if (subscribed == NONE) {
         /* Can't find a format for events that we know about */
         char mybuf[200];
         if (!ast_strlen_zero(unknown_acceptheader)) {
            snprintf(mybuf, sizeof(mybuf), "489 Bad Event (format %s)", unknown_acceptheader);
         } else {
            snprintf(mybuf, sizeof(mybuf), "489 Bad Event");
         }
         transmit_response(p, mybuf, req);
         ast_log(LOG_WARNING,"SUBSCRIBE failure: unrecognized format:"
            "'%s' pvt: subscribed: %d, stateid: %d, laststate: %d,"
            "dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n",
            unknown_acceptheader,
            (int)p->subscribed,
            p->stateid,
            p->laststate,
            p->dialogver,
            p->subscribecontext,
            p->subscribeuri);
         pvt_set_needdestroy(p, "unrecognized format");
         return 0;
      } else {
         p->subscribed = subscribed;
      }
   } else if (!strcmp(event, "message-summary")) {
      int start = 0;
      int found_supported = 0;
      const char *acceptheader;

      acceptheader = __get_header(req, "Accept", &start);
      while (!found_supported && !ast_strlen_zero(acceptheader)) {
         found_supported = strcmp(acceptheader, "application/simple-message-summary") ? 0 : 1;
         if (!found_supported && (option_debug > 2)) {
            ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
         }
         acceptheader = __get_header(req, "Accept", &start);
      }
      if (start && !found_supported) {
         /* Format requested that we do not support */
         transmit_response(p, "406 Not Acceptable", req);
         ast_debug(2, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
         pvt_set_needdestroy(p, "unknown format");
         if (authpeer)
            unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 3)");
         return 0;
      }
      /* Looks like they actually want a mailbox status
        This version of Asterisk supports mailbox subscriptions
        The subscribed URI needs to exist in the dial plan
        In most devices, this is configurable to the voicemailmain extension you use
      */
      if (!authpeer || AST_LIST_EMPTY(&authpeer->mailboxes)) {
         transmit_response(p, "404 Not found (no mailbox)", req);
         pvt_set_needdestroy(p, "received 404 response");
         ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
         if (authpeer)
            unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 4)");
         return 0;
      }

      p->subscribed = MWI_NOTIFICATION;
      if (ast_test_flag(&authpeer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)) {
         add_peer_mwi_subs(authpeer);
      }
      if (authpeer->mwipvt && authpeer->mwipvt != p) {   /* Destroy old PVT if this is a new one */
         /* We only allow one subscription per peer */
         dialog_unlink_all(authpeer->mwipvt, TRUE, TRUE);
         authpeer->mwipvt = dialog_unref(authpeer->mwipvt, "unref dialog authpeer->mwipvt");
         /* sip_destroy(authpeer->mwipvt); */
      }
      if (authpeer->mwipvt)
         dialog_unref(authpeer->mwipvt, "Unref previously stored mwipvt dialog pointer");
      authpeer->mwipvt = dialog_ref(p, "setting peers' mwipvt to p");      /* Link from peer to pvt UH- should this be dialog_ref()? */
      if (p->relatedpeer)
         unref_peer(p->relatedpeer, "Unref previously stored relatedpeer ptr");
      p->relatedpeer = ref_peer(authpeer, "setting dialog's relatedpeer pointer");  /* already refcounted...Link from pvt to peer UH- should this be dialog_ref()? */
      /* Do not release authpeer here */
   } else if (!strcmp(event, "call-completion")) {
      handle_cc_subscribe(p, req);
   } else { /* At this point, Asterisk does not understand the specified event */
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event);
      pvt_set_needdestroy(p, "unknown event package");
      if (authpeer)
         unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 5)");
      return 0;
   }

   /* Add subscription for extension state from the PBX core */
   if (p->subscribed != MWI_NOTIFICATION  && p->subscribed != CALL_COMPLETION && !resubscribe) {
      if (p->stateid > -1) {
         ast_extension_state_del(p->stateid, cb_extensionstate);
         /* we need to dec the refcount, now that the extensionstate is removed */
         dialog_unref(p, "the extensionstate containing this dialog ptr was deleted");
      }
      p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, dialog_ref(p,"copying dialog ptr into extension state struct"));
   }

   if (!req->ignore && p)
      p->lastinvite = seqno;
   if (p && !p->needdestroy) {
      p->expiry = atoi(get_header(req, "Expires"));

      /* check if the requested expiry-time is within the approved limits from sip.conf */
      if (p->expiry > max_expiry) {
         p->expiry = max_expiry;
      } else if (p->expiry < min_expiry && p->expiry > 0) {
         transmit_response_with_minexpires(p, "423 Interval too small", req);
         ast_log(LOG_WARNING, "Received subscription for extension \"%s\" context \"%s\" "
            "with Expire header less that 'minexpire' limit. Received \"Expire: %d\" min is %d\n",
            p->exten, p->context, p->expiry, min_expiry);
         p->expiry = min_expiry;
         pvt_set_needdestroy(p, "Expires is less that the min expires allowed. ");
         return 0;
      }

      if (sipdebug) {
         if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) {
            ast_debug(2, "Adding subscription for mailbox notification - peer %s\n", p->relatedpeer->name);
         } else if (p->subscribed == CALL_COMPLETION) {
            ast_debug(2, "Adding CC subscription for peer %s\n", p->username);
         } else {
            ast_debug(2, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username);
         }
      }
      if (p->autokillid > -1 && sip_cancel_destroy(p))   /* Remove subscription expiry for renewals */
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (p->expiry > 0)
         sip_scheddestroy(p, (p->expiry + 10) * 1000);   /* Set timer for destruction of call at expiration */

      if (p->subscribed == MWI_NOTIFICATION) {
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         if (p->relatedpeer) {   /* Send first notification */
            ao2_lock(p->relatedpeer); /* was WRLOCK */
            sip_send_mwi_to_peer(p->relatedpeer, NULL, 0);
            ao2_unlock(p->relatedpeer);
         }
      } else if (p->subscribed != CALL_COMPLETION) {
         struct sip_pvt *p_old;

         if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) {

            ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_sockaddr_stringify(&p->sa));
            transmit_response(p, "404 Not found", req);
            pvt_set_needdestroy(p, "no extension for SUBSCRIBE");
            return 0;
         }
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         transmit_state_notify(p, firststate, 1, FALSE); /* Send first notification */
         append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));
         /* hide the 'complete' exten/context in the refer_to field for later display */
         ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);

         /* remove any old subscription from this peer for the same exten/context,
         as the peer has obviously forgotten about it and it's wasteful to wait
         for it to expire and send NOTIFY messages to the peer only to have them
         ignored (or generate errors)
         */
         i = ao2_iterator_init(dialogs, 0);
         while ((p_old = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
            if (p_old == p) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            if (p_old->initreq.method != SIP_SUBSCRIBE) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            if (p_old->subscribed == NONE) {
               ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
               continue;
            }
            sip_pvt_lock(p_old);
            if (!strcmp(p_old->username, p->username)) {
               if (!strcmp(p_old->exten, p->exten) &&
                   !strcmp(p_old->context, p->context)) {
                  pvt_set_needdestroy(p_old, "replacing subscription");
                  sip_pvt_unlock(p_old);
                  ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before break");
                  break;
               }
            }
            sip_pvt_unlock(p_old);
            ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next");
         }
         ao2_iterator_destroy(&i);
      }
      if (!p->expiry) {
         pvt_set_needdestroy(p, "forcing expiration");
      }
   }
   return 1;
}

static int handle_request_update	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

bare-bones support for SIP UPDATE

XXX This is not even close to being RFC 3311-compliant. We don't advertise that we support the UPDATE method, so no one should ever try sending us an UPDATE anyway. However, Asterisk can send an UPDATE to change connected line information, so we need to be prepared to handle this. The way we distinguish such an UPDATE is through the X-Asterisk-rpid-update header.

Actually updating the media session may be some future work.

Definition at line 20751 of file chan_sip.c.

References ast_channel_queue_connected_line_update(), AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, ast_party_connected_line_init(), ast_strlen_zero(), connected, get_header(), get_rpid(), ast_party_connected_line::id, ast_set_party_connected_line::id, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, ast_party_name::presentation, ast_party_number::presentation, ast_party_connected_line::source, ast_party_name::str, ast_party_number::str, ast_party_id::tag, transmit_response(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_incoming().

{
   if (ast_strlen_zero(get_header(req, "X-Asterisk-rpid-update"))) {
      transmit_response(p, "501 Method Not Implemented", req);
      return 0;
   }
   if (get_rpid(p, req)) {
      struct ast_party_connected_line connected;
      struct ast_set_party_connected_line update_connected;
      ast_party_connected_line_init(&connected);
      memset(&update_connected, 0, sizeof(update_connected));
      if (p->cid_num) {
         update_connected.id.number = 1;
         connected.id.number.valid = 1;
         connected.id.number.str = (char *) p->cid_num;
         connected.id.number.presentation = p->callingpres;
      }
      if (p->cid_name) {
         update_connected.id.name = 1;
         connected.id.name.valid = 1;
         connected.id.name.str = (char *) p->cid_name;
         connected.id.name.presentation = p->callingpres;
      }
      connected.id.tag = (char *) p->cid_tag;
      connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
      ast_channel_queue_connected_line_update(p->owner, &connected, &update_connected);
   }
   transmit_response(p, "200 OK", req);
   return 0;
}

static void handle_response	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle SIP response in dialogue.

Note:

only called by handle_incoming

Definition at line 19546 of file chan_sip.c.

References __sip_ack(), __sip_semi_ack(), ao2_callback, append_history, AST_CAUSE_PROTOCOL_ERROR, AST_CC_CCBS, ast_cc_monitor_failed(), ast_channel_set_redirecting(), ast_clear_flag, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, ast_debug, ast_log(), ast_party_redirecting_free(), ast_party_redirecting_init(), ast_queue_control(), ast_queue_hangup_with_cause(), ast_set_flag, ast_skip_blanks(), ast_skip_nonblanks(), ast_sockaddr_stringify(), ast_strdupa, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, ast_verbose, cb_extensionstate(), change_redirecting_information(), do_proxy_auth(), FALSE, find_sdp(), find_sip_method(), find_sip_monitor_instance_by_subscription_pvt(), get_header(), gettag(), handle_response_invite(), handle_response_notify(), handle_response_peerpoke(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), handle_response_update(), hangup_sip2cause(), ast_channel::hangupcause, LOG_NOTICE, LOG_WARNING, mark_method_allowed(), mark_method_unallowed(), process_sdp(), pvt_set_needdestroy(), rh, sip_alreadygone(), sip_cancel_destroy(), sip_handle_cc(), sip_methods, stop_media_flows(), text, transmit_request(), TRUE, and update_redirecting().

{
   struct ast_channel *owner;
   int sipmethod;
   int res = 1;
   const char *c = get_header(req, "Cseq");
   /* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
   char *c_copy = ast_strdupa(c);
   /* Skip the Cseq and its subsequent spaces */
   const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));

   if (!msg)
      msg = "";

   sipmethod = find_sip_method(msg);

   owner = p->owner;
   if (owner) {
      const char *rp = NULL, *rh = NULL;

      owner->hangupcause = 0;
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON) && (rh = get_header(req, "Reason"))) {
         rh = ast_skip_blanks(rh);
         if (!strncasecmp(rh, "Q.850", 5)) {
            rp = strstr(rh, "cause=");
            if (rp && sscanf(rp + 6, "%30d", &owner->hangupcause) == 1) {
               owner->hangupcause &= 0x7f;
               if (req->debug)
                  ast_verbose("Using Reason header for cause code: %d\n", owner->hangupcause);
            }
         }
      }

      if (!owner->hangupcause)
         owner->hangupcause = hangup_sip2cause(resp);
   }

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      int ack_res = FALSE;

      /* Acknowledge whatever it is destined for */
      if ((resp >= 100) && (resp <= 199)) {
         /* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
         if (sipmethod == SIP_INVITE) {
            ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
         }
      } else {
         ack_res = __sip_ack(p, seqno, 0, sipmethod);
      }

      if (ack_res == FALSE) {
         /* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
         if (sipmethod == SIP_INVITE && resp >= 200) {
            transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
         }

         append_history(p, "Ignore", "Ignoring this retransmit\n");
         return;
      }
   }

   /* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
   if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite)
      p->pendinginvite = 0;

   /* Get their tag if we haven't already */
   if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
      char tag[128];

      gettag(req, "To", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   }
   /* This needs to be configurable on a channel/peer level,
      not mandatory for all communication. Sadly enough, NAT implementations
      are not so stable so we can always rely on these headers.
      Temporarily disabled, while waiting for fix.
      Fix assigned to Rizzo :-)
   */
   /* check_via_response(p, req); */

   /* RFC 3261 Section 15 specifies that if we receive a 408 or 481
    * in response to a BYE, then we should end the current dialog
    * and session.  It is known that at least one phone manufacturer
    * potentially will send a 404 in response to a BYE, so we'll be
    * liberal in what we accept and end the dialog and session if we
    * receive any of those responses to a BYE.
    */
   if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
      pvt_set_needdestroy(p, "received 4XX response to a BYE");
      return;
   }

   if (p->relatedpeer && p->method == SIP_OPTIONS) {
      /* We don't really care what the response is, just that it replied back.
         Well, as long as it's not a 100 response...  since we might
         need to hang around for something more "definitive" */
      if (resp != 100)
         handle_response_peerpoke(p, resp, req);
   } else if (sipmethod == SIP_REFER && resp >= 200) {
      handle_response_refer(p, resp, rest, req, seqno);
   } else if (sipmethod == SIP_PUBLISH) {
      /* SIP PUBLISH transcends this morass of doodoo and instead
       * we just always call the response handler. Good gravy!
       */
      handle_response_publish(p, resp, rest, req, seqno);
   } else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      switch(resp) {
      case 100:   /* 100 Trying */
      case 101:   /* 101 Dialog establishment */
      case 183:   /* 183 Session Progress */
      case 180:   /* 180 Ringing */
      case 182:   /* 182 Queued */
      case 181:   /* 181 Call Is Being Forwarded */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 200:   /* 200 OK */
         p->authtries = 0; /* Reset authentication counter */
         if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
            /* We successfully transmitted a message
               or a video update request in INFO */
            /* Nothing happens here - the message is inside a dialog */
         } else if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_NOTIFY) {
            handle_response_notify(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_REGISTER) {
            res = handle_response_register(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_SUBSCRIBE) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            handle_response_subscribe(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {     /* Ok, we're ready to go */
            pvt_set_needdestroy(p, "received 200 response");
            ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         }
         break;
      case 401: /* Not www-authorized on SIP method */
      case 407: /* Proxy auth required */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_NOTIFY)
            handle_response_notify(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_UPDATE) {
            handle_response_update(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            if (p->options)
               p->options->auth_type = resp;
            if (ast_strlen_zero(p->authname)) {
               ast_log(LOG_WARNING, "Asked to authenticate %s, to %s but we have no matching peer!\n",
                     msg, ast_sockaddr_stringify(&p->recv));
               pvt_set_needdestroy(p, "unable to authenticate BYE");
            } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp,  sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
               pvt_set_needdestroy(p, "failed to authenticate BYE");
            }
         } else {
            ast_log(LOG_WARNING, "Got authentication request (%d) on %s to '%s'\n", resp, sip_methods[sipmethod].text, get_header(req, "To"));
            pvt_set_needdestroy(p, "received 407 response");
         }
         break;
      case 403: /* Forbidden - we failed authentication */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else {
            ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg);
            pvt_set_needdestroy(p, "received 403 response");
         }
         break;
      case 404: /* Not found */
         if (p->registry && sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (owner)
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         break;
      case 423: /* Interval too brief */
         if (sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         break;
      case 408: /* Request timeout - terminate dialog */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REGISTER)
            res = handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "received 408 response");
            ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
         } else {
            if (owner)
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            pvt_set_needdestroy(p, "received 408 response");
         }
         break;

      case 422: /* Session-Timers: Session Interval Too Small */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         }
         break;

      case 481: /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_SUBSCRIBE) {
            handle_response_subscribe(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            /* The other side has no transaction to bye,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else if (sipmethod == SIP_CANCEL) {
            /* The other side has no transaction to cancel,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else {
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
            /* Guessing that this is not an important request */
         }
         break;
      case 487:
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 415: /* Unsupported media type */
      case 488: /* Not acceptable here - codec error */
      case 606: /* Not Acceptable */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 491: /* Pending */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else {
            ast_debug(1, "Got 491 on %s, unsupported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
            pvt_set_needdestroy(p, "received 491 response");
         }
         break;
      case 405:
      case 501: /* Not Implemented */
         mark_method_unallowed(&p->allowed_methods, sipmethod);
         if (p->relatedpeer) {
            mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
         }
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else
            ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_sockaddr_stringify(&p->sa), msg);
         break;
         /* Fallthrough */
      default:
         if ((resp >= 300) && (resp < 700)) {
            /* Fatal response */
            if ((resp != 487))
               ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
   
            if (sipmethod == SIP_INVITE)
               stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */

            /* XXX Locking issues?? XXX */
            switch(resp) {
            case 300: /* Multiple Choices */
            case 301: /* Moved permanently */
            case 302: /* Moved temporarily */
            case 305: /* Use Proxy */
            if (p->owner) {
               struct ast_party_redirecting redirecting;
               struct ast_set_party_redirecting update_redirecting;

               ast_party_redirecting_init(&redirecting);
               change_redirecting_information(p, req, &redirecting,
                  &update_redirecting, TRUE);
               ast_channel_set_redirecting(p->owner, &redirecting,
                  &update_redirecting);
               ast_party_redirecting_free(&redirecting);
            }
               /* Fall through */
            case 486: /* Busy here */
            case 600: /* Busy everywhere */
            case 603: /* Decline */
               if (p->owner) {
                  sip_handle_cc(p, req, AST_CC_CCBS);
                  ast_queue_control(p->owner, AST_CONTROL_BUSY);
               }
               break;
            case 482: /* Loop Detected */
            case 480: /* Temporarily Unavailable */
            case 404: /* Not Found */
            case 410: /* Gone */
            case 400: /* Bad Request */
            case 500: /* Server error */
               if (sipmethod == SIP_SUBSCRIBE) {
                  handle_response_subscribe(p, resp, rest, req, seqno);
                  break;
               }
               /* Fall through */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server Timeout */
               if (owner)
                  ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
               break;
            default:
               /* Send hangup */ 
               if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
                  ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
               break;
            }
            /* ACK on invite */
            if (sipmethod == SIP_INVITE)
               transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
            if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO)
               sip_alreadygone(p);
            if (!p->owner) {
               pvt_set_needdestroy(p, "transaction completed");
            }
         } else if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               if (find_sdp(req))
                  process_sdp(p, req, SDP_T38_NONE);
               if (p->owner) {
                  /* Queue a progress frame */
                  ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
               }
            }
         } else
            ast_log(LOG_NOTICE, "Don't know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_sockaddr_stringify(&p->sa));
      }
   } else { 
      /* Responses to OUTGOING SIP requests on INCOMING calls
         get handled here. As well as out-of-call message responses */
      if (req->debug)
         ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);

      if (sipmethod == SIP_INVITE && resp == 200) {
         /* Tags in early session is replaced by the tag in 200 OK, which is
         the final reply to our INVITE */
         char tag[128];

         gettag(req, "To", tag, sizeof(tag));
         ast_string_field_set(p, theirtag, tag);
      }

      if (sipmethod == SIP_SUBSCRIBE && resp >= 400) {
         struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances,
               0, find_sip_monitor_instance_by_subscription_pvt, p);
         if (monitor_instance) {
            ast_cc_monitor_failed(monitor_instance->core_id, monitor_instance->device_name,
                  "Received error response to our SUBSCRIBE");
            return;
         }
      }

      switch(resp) {
      case 200:
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_CANCEL) {
            ast_debug(1, "Got 200 OK on CANCEL\n");

            /* Wait for 487, then destroy */
         } else if (sipmethod == SIP_NOTIFY) {
            /* They got the notify, this is the end */
            if (p->owner) {
               if (p->refer) {
                  ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
               } else
                  ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
               /* ast_queue_hangup(p->owner); Disabled */
            } else {
               if (!p->subscribed && !p->refer) {
                  pvt_set_needdestroy(p, "transaction completed");
               }
               if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
                  /* Ready to send the next state we have on queue */
                  ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
                  cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
               }
            }
         } else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "transaction completed");
         } else if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
            /* We successfully transmitted a message or
               a video update request in INFO */
            ;
         }
         break;
      case 401:   /* www-auth */
      case 407:
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
               pvt_set_needdestroy(p, "failed to authenticate BYE");
            }
         }
         break;
      case 481:   /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            /* Re-invite failed */
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "received 481 response");
         } else if (sipdebug) {
            ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
         }
         break;
      case 501: /* Not Implemented */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      default: /* Errors without handlers */
         if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {   /* re-invite */
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
            }
         }
         if ((resp >= 300) && (resp < 700)) {
            if ((resp != 487))
               ast_verb(3, "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
            switch(resp) {
            case 415: /* Unsupported media type */
            case 488: /* Not acceptable here - codec error */
            case 603: /* Decline */
            case 500: /* Server error */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server timeout */

               /* re-invite failed */
               if (sipmethod == SIP_INVITE && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               break;
            }
         }
         break;
      }
   }
}

static void handle_response_invite	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle SIP response to INVITE dialogue.

Definition at line 18703 of file chan_sip.c.

References append_history, AST_CAUSE_NORMAL_CLEARING, AST_CC_CCNR, ast_channel_queue_connected_line_update(), ast_channel_queue_redirecting_update(), AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, AST_CONTROL_ANSWER, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_debug, ast_log(), ast_null_frame, ast_party_connected_line_init(), ast_party_redirecting_free(), ast_party_redirecting_init(), ast_queue_control(), ast_queue_frame(), ast_queue_hangup_with_cause(), ast_random(), ast_rtp_instance_activate(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_set_hangupsource(), ast_setstate(), AST_STATE_RINGING, AST_STATE_UP, ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_route(), change_redirecting_information(), change_t38_state(), check_pendings(), connected, do_proxy_auth(), EVENT_FLAG_SYSTEM, FALSE, find_sdp(), get_header(), get_rpid(), ast_party_connected_line::id, ast_set_party_connected_line::id, LOG_NOTICE, LOG_WARNING, manager_event, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_ok_contact(), parse_session_expires(), ast_party_name::presentation, ast_party_number::presentation, proc_422_rsp(), process_sdp(), pvt_set_needdestroy(), set_address_from_contact(), set_pvt_allowed_methods(), sip_alreadygone(), sip_cancel_destroy(), sip_cfg, sip_handle_cc(), sip_reinvite_retry(), sip_scheddestroy(), ast_party_connected_line::source, st_get_mode(), start_session_timer(), ast_party_name::str, ast_party_number::str, ast_party_id::tag, transmit_reinvite_with_sdp(), transmit_request(), TRUE, update_call_counter(), update_redirecting(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_response().

{
   int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
   int res = 0;
   int xmitres = 0;
   int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
   char *p_hdrval;
   int rtn;
   struct ast_party_connected_line connected;
   struct ast_set_party_connected_line update_connected;

   if (reinvite)
      ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
   else
      ast_debug(4, "SIP response %d to standard invite\n", resp);

   if (p->alreadygone) { /* This call is already gone */
      ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
      return;
   }

   /* Acknowledge sequence number - This only happens on INVITE from SIP-call */
   /* Don't auto congest anymore since we've gotten something useful back */
   AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));

   /* RFC3261 says we must treat every 1xx response (but not 100)
      that we don't recognize as if it was 183.
   */
   if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
      resp = 183;

   /* Any response between 100 and 199 is PROCEEDING */
   if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
      p->invitestate = INV_PROCEEDING;

   /* Final response, not 200 ? */
   if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
      p->invitestate = INV_COMPLETED;
   
   /* Final response, clear out pending invite */
   if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
      p->pendinginvite = 0;

   /* If this is a response to our initial INVITE, we need to set what we can use
    * for this peer.
    */
   if (!reinvite) {
      set_pvt_allowed_methods(p, req);
   }     

   switch (resp) {
   case 100:   /* Trying */
   case 101:   /* Dialog establishment */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      check_pendings(p);
      break;

   case 180:   /* 180 Ringing */
   case 182:       /* 182 Queued */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (!req->ignore && p->owner) {
         if (get_rpid(p, req)) {
            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));
            if (p->cid_num) {
               update_connected.id.number = 1;
               connected.id.number.valid = 1;
               connected.id.number.str = (char *) p->cid_num;
               connected.id.number.presentation = p->callingpres;
            }
            if (p->cid_name) {
               update_connected.id.name = 1;
               connected.id.name.valid = 1;
               connected.id.name.str = (char *) p->cid_name;
               connected.id.name.presentation = p->callingpres;
            }
            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         sip_handle_cc(p, req, AST_CC_CCNR);
         ast_queue_control(p->owner, AST_CONTROL_RINGING);
         if (p->owner->_state != AST_STATE_UP) {
            ast_setstate(p->owner, AST_STATE_RINGING);
         }
      }
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED)
            p->invitestate = INV_EARLY_MEDIA;
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame only if we have SDP in 180 or 182 */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
         ast_rtp_instance_activate(p->rtp);
      }
      check_pendings(p);
      break;

   case 181:   /* Call Is Being Forwarded */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (!req->ignore && p->owner) {
         struct ast_party_redirecting redirecting;
         struct ast_set_party_redirecting update_redirecting;

         ast_party_redirecting_init(&redirecting);
         memset(&update_redirecting, 0, sizeof(update_redirecting));
         change_redirecting_information(p, req, &redirecting, &update_redirecting,
            FALSE);
         ast_channel_queue_redirecting_update(p->owner, &redirecting,
            &update_redirecting);
         ast_party_redirecting_free(&redirecting);
         sip_handle_cc(p, req, AST_CC_CCNR);
      }
      check_pendings(p);
      break;

   case 183:   /* Session progress */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (!req->ignore && p->owner) {
         if (get_rpid(p, req)) {
            /* Queue a connected line update */
            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));
            if (p->cid_num) {
               update_connected.id.number = 1;
               connected.id.number.valid = 1;
               connected.id.number.str = (char *) p->cid_num;
               connected.id.number.presentation = p->callingpres;
            }
            if (p->cid_name) {
               update_connected.id.name = 1;
               connected.id.name.valid = 1;
               connected.id.name.str = (char *) p->cid_name;
               connected.id.name.presentation = p->callingpres;
            }
            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         sip_handle_cc(p, req, AST_CC_CCNR);
      }
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED)
            p->invitestate = INV_EARLY_MEDIA;
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
         ast_rtp_instance_activate(p->rtp);
      } else {
         /* Alcatel PBXs are known to send 183s with no SDP after sending
          * a 100 Trying response. We're just going to treat this sort of thing
          * the same as we would treat a 180 Ringing
          */
         if (!req->ignore && p->owner) {
            ast_queue_control(p->owner, AST_CONTROL_RINGING);
         }
      }
      check_pendings(p);
      break;

   case 200:   /* 200 OK on invite - someone's answering our call */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      p->authtries = 0;
      if (find_sdp(req)) {
         if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
            if (!reinvite)
               /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
               /* For re-invites, we try to recover */
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
         ast_rtp_instance_activate(p->rtp);
      }

      if (!req->ignore && p->owner && (get_rpid(p, req) || !reinvite)) {
         /* Queue a connected line update */
         ast_party_connected_line_init(&connected);
         memset(&update_connected, 0, sizeof(update_connected));
         if (p->cid_num) {
            update_connected.id.number = 1;
            connected.id.number.valid = 1;
            connected.id.number.str = (char *) p->cid_num;
            connected.id.number.presentation = p->callingpres;
         }
         if (p->cid_name) {
            update_connected.id.name = 1;
            connected.id.name.valid = 1;
            connected.id.name.str = (char *) p->cid_name;
            connected.id.name.presentation = p->callingpres;
         }
         connected.id.tag = (char *) p->cid_tag;
         connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
         ast_channel_queue_connected_line_update(p->owner, &connected,
            &update_connected);
      }

      /* Parse contact header for continued conversation */
      /* When we get 200 OK, we know which device (and IP) to contact for this call */
      /* This is important when we have a SIP proxy between us and the phone */
      if (outgoing) {
         update_call_counter(p, DEC_CALL_RINGING);
         parse_ok_contact(p, req);
         /* Save Record-Route for any later requests we make on this dialogue */
         if (!reinvite)
            build_route(p, req, 1);

         if(set_address_from_contact(p)) {
            /* Bad contact - we don't know how to reach this device */
            /* We need to ACK, but then send a bye */
            if (!p->route && !req->ignore)
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
         }

      }

      if (!req->ignore && p->owner) {
         if (!reinvite) {
            ast_queue_control(p->owner, AST_CONTROL_ANSWER);
            if (sip_cfg.callevents)
               manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
                  "Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
                  p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
         } else { /* RE-invite */
            ast_queue_frame(p->owner, &ast_null_frame);
         }
      } else {
          /* It's possible we're getting an 200 OK after we've tried to disconnect
              by sending CANCEL */
         /* First send ACK, then send bye */
         if (!req->ignore)
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
      }

      /* Check for Session-Timers related headers */
      if (st_get_mode(p) != SESSION_TIMER_MODE_REFUSE && p->outgoing_call == TRUE && !reinvite) {
         p_hdrval = (char*)get_header(req, "Session-Expires");
         if (!ast_strlen_zero(p_hdrval)) {
            /* UAS supports Session-Timers */
            enum st_refresher tmp_st_ref = SESSION_TIMER_REFRESHER_AUTO;
            int tmp_st_interval = 0;
            rtn = parse_session_expires(p_hdrval, &tmp_st_interval, &tmp_st_ref);
            if (rtn != 0) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            }
            if (tmp_st_ref == SESSION_TIMER_REFRESHER_UAC ||
               tmp_st_ref == SESSION_TIMER_REFRESHER_UAS) {
               p->stimer->st_ref = tmp_st_ref;
            }
            if (tmp_st_interval) {
               p->stimer->st_interval = tmp_st_interval;
            }
            p->stimer->st_active = TRUE;
            p->stimer->st_active_peer_ua = TRUE;
            start_session_timer(p);
         } else {
            /* UAS doesn't support Session-Timers */
            if (st_get_mode(p) == SESSION_TIMER_MODE_ORIGINATE) {
               p->stimer->st_ref = SESSION_TIMER_REFRESHER_UAC;
               p->stimer->st_active_peer_ua = FALSE;
               start_session_timer(p);
            }
         }
      }


      /* If I understand this right, the branch is different for a non-200 ACK only */
      p->invitestate = INV_TERMINATED;
      ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
      check_pendings(p);
      break;

   case 407: /* Proxy authentication */
   case 401: /* Www auth */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->options)
         p->options->auth_type = resp;

      /* Then we AUTH */
      ast_string_field_set(p, theirtag, NULL);  /* forget their old tag, so we don't match tags when getting response */
      if (!req->ignore) {
         if (p->authtries < MAX_AUTHTRIES)
            p->invitestate = INV_CALLING;
         if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
            ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
            pvt_set_needdestroy(p, "failed to authenticate on INVITE");
            sip_alreadygone(p);
            if (p->owner)
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         }
      }
      break;

   case 403: /* Forbidden */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
      if (!req->ignore && p->owner) {
         ast_set_hangupsource(p->owner, p->owner->name, 0);
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      }
      pvt_set_needdestroy(p, "received 403 response");
      sip_alreadygone(p);
      break;

   case 404: /* Not found */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         ast_set_hangupsource(p->owner, p->owner->name, 0);
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      }
      sip_alreadygone(p);
      break;

   case 408: /* Request timeout */
   case 481: /* Call leg does not exist */
      /* Could be REFER caused INVITE with replaces */
      ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      break;

   case 422: /* Session-Timers: Session interval too small */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_string_field_set(p, theirtag, NULL);
      proc_422_rsp(p, req);
      break;

   case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
      ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      break;

      

   case 487: /* Cancelled transaction */
      /* We have sent CANCEL on an outbound INVITE
         This transaction is already scheduled to be killed by sip_hangup().
      */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
      } else if (!req->ignore) {
         update_call_counter(p, DEC_CALL_LIMIT);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
         pvt_set_needdestroy(p, "received 487 response");
         sip_alreadygone(p);
      }
      break;
   case 415: /* Unsupported media type */
   case 488: /* Not acceptable here */
   case 606: /* Not Acceptable */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
         change_t38_state(p, T38_DISABLED);
         /* Try to reset RTP timers */
         //ast_rtp_set_rtptimers_onhold(p->rtp);

         /* Trigger a reinvite back to audio */
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else {
         /* We can't set up this call, so give up */
         if (p->owner && !req->ignore)
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         pvt_set_needdestroy(p, "received 488 response");
         /* If there's no dialog to end, then mark p as already gone */
         if (!reinvite)
            sip_alreadygone(p);
      }
      break;
   case 491: /* Pending */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         if (p->owner->_state != AST_STATE_UP) {
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            pvt_set_needdestroy(p, "received 491 response");
         } else {
            /* This is a re-invite that failed. */
            /* Reset the flag after a while
             */
            int wait;
            /* RFC 3261, if owner of call, wait between 2.1 to 4 seconds,
             * if not owner of call, wait 0 to 2 seconds */
            if (p->outgoing_call) {
               wait = 2100 + ast_random() % 2000;
            } else {
               wait = ast_random() % 2000;
            }
            p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, dialog_ref(p, "passing dialog ptr into sched structure based on waitid for sip_reinvite_retry."));
            ast_log(LOG_WARNING, "just did sched_add waitid(%d) for sip_reinvite_retry for dialog %s in handle_response_invite\n", p->waitid, p->callid);
            ast_debug(2, "Reinvite race. Waiting %d secs before retry\n", wait);
         }
      }
      break;

   case 405: /* Not allowed */
   case 501: /* Not implemented */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      break;
   }
   if (xmitres == XMIT_ERROR)
      ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
}

static void handle_response_notify	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 19127 of file chan_sip.c.

References AST_CAUSE_NORMAL_UNSPECIFIED, ast_clear_flag, ast_debug, ast_log(), ast_queue_hangup_with_cause(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, cb_extensionstate(), do_proxy_auth(), get_header(), LOG_NOTICE, LOG_WARNING, NONE, and pvt_set_needdestroy().

Referenced by handle_response().

{
   switch (resp) {
   case 200:   /* Notify accepted */
      /* They got the notify, this is the end */
      if (p->owner) {
         if (!p->refer) {
            ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
            ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
         } else {
            ast_debug(4, "Got OK on REFER Notify message\n");
         }
      } else {
         if (p->subscribed == NONE) {
            ast_debug(4, "Got 200 accepted on NOTIFY\n");
            pvt_set_needdestroy(p, "received 200 response");
         }
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
            /* Ready to send the next state we have on queue */
            ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
            cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
         }
      }
      break;
   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (!p->notify) {
         break; /* Only device notify can use NOTIFY auth */
      }
      ast_string_field_set(p, theirtag, NULL);
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate NOTIFY to %s but we have no matching peer or realm auth!\n", ast_sockaddr_stringify(&p->recv));
         pvt_set_needdestroy(p, "unable to authenticate NOTIFY");
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_NOTIFY, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on NOTIFY to '%s'\n", get_header(&p->initreq, "From"));
         pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
      }
      break;
   }
}

static void handle_response_peerpoke	(	struct sip_pvt *	p,
		int	resp,
		struct sip_request *	req
	)			[static]

Handle qualification responses (OPTIONS).

Definition at line 19476 of file chan_sip.c.

References ast_check_realtime(), AST_DEVICE_UNKNOWN, ast_devstate_changed(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_tvdiff_ms(), ast_tvnow(), ast_update_realtime(), DEFAULT_FREQ_NOTOK, EVENT_FLAG_SYSTEM, LOG_NOTICE, manager_event, pvt_set_needdestroy(), ref_peer(), register_peer_exten(), SENTINEL, sip_cfg, sip_poke_peer_s(), TRUE, and unref_peer().

Referenced by handle_response().

{
   struct sip_peer *peer = /* ref_peer( */ p->relatedpeer /* , "bump refcount on p, as it is being used in this function(handle_response_peerpoke)")*/ ; /* hope this is already refcounted! */
   int statechanged, is_reachable, was_reachable;
   int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);

   /*
    * Compute the response time to a ping (goes in peer->lastms.)
    * -1 means did not respond, 0 means unknown,
    * 1..maxms is a valid response, >maxms means late response.
    */
   if (pingtime < 1) /* zero = unknown, so round up to 1 */
      pingtime = 1;

   /* Now determine new state and whether it has changed.
    * Use some helper variables to simplify the writing
    * of the expressions.
    */
   was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
   is_reachable = pingtime <= peer->maxms;
   statechanged = peer->lastms == 0 /* yes, unknown before */
      || was_reachable != is_reachable;

   peer->lastms = pingtime;
   peer->call = dialog_unref(peer->call, "unref dialog peer->call");
   if (statechanged) {
      const char *s = is_reachable ? "Reachable" : "Lagged";
      char str_lastms[20];
      snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);

      ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
         peer->name, s, pingtime, peer->maxms);
      ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
      if (sip_cfg.peer_rtupdate) {
         ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
         "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
         peer->name, s, pingtime);
      if (is_reachable && sip_cfg.regextenonqualify)
         register_peer_exten(peer, TRUE);
   }

   pvt_set_needdestroy(p, "got OPTIONS response");

   /* Try again eventually */
   AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
         is_reachable ? peer->qualifyfreq : DEFAULT_FREQ_NOTOK,
         sip_poke_peer_s, peer,
         unref_peer(_data, "removing poke peer ref"),
         unref_peer(peer, "removing poke peer ref"),
         ref_peer(peer, "adding poke peer ref"));
}

static void handle_response_publish	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 18652 of file chan_sip.c.

References ast_assert, ast_copy_string(), ast_log(), ast_string_field_set, ast_strlen_zero(), do_proxy_auth(), get_header(), LOG_NOTICE, mark_method_unallowed(), pvt_set_needdestroy(), and sip_alreadygone().

Referenced by handle_response().

{
   struct sip_epa_entry *epa_entry = p->epa_entry;
   const char *etag = get_header(req, "Sip-ETag");

   ast_assert(epa_entry != NULL);

   if (resp == 401 || resp == 407) {
      ast_string_field_set(p, theirtag, NULL);
      if (p->options) {
         p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
      }
      if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_PUBLISH, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on PUBLISH to '%s'\n", get_header(&p->initreq, "From"));
         pvt_set_needdestroy(p, "Failed to authenticate on PUBLISH");
         sip_alreadygone(p);
      }
      return;
   }

   if (resp == 501 || resp == 405) {
      mark_method_unallowed(&p->allowed_methods, SIP_PUBLISH);
   }

   if (resp == 200) {
      p->authtries = 0;
      /* If I've read section 6, item 6 of RFC 3903 correctly,
       * an ESC will only generate a new etag when it sends a 200 OK
       */
      if (!ast_strlen_zero(etag)) {
         ast_copy_string(epa_entry->entity_tag, etag, sizeof(epa_entry->entity_tag));
      }
      /* The nominal case. Everything went well. Everybody is happy.
       * Each EPA will have a specific action to take as a result of this
       * development, so ... callbacks!
       */
      if (epa_entry->static_data->handle_ok) {
         epa_entry->static_data->handle_ok(p, req, epa_entry);
      }
   } else {
      /* Rather than try to make individual callbacks for each error
       * type, there is just a single error callback. The callback
       * can distinguish between error messages and do what it needs to
       */
      if (epa_entry->static_data->handle_error) {
         epa_entry->static_data->handle_error(p, resp, req, epa_entry);
      }
   }
}

static void handle_response_refer	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 19232 of file chan_sip.c.

References AST_CONTROL_CONGESTION, AST_CONTROL_TRANSFER, ast_debug, ast_log(), ast_queue_control(), ast_queue_control_data(), ast_sockaddr_stringify(), ast_strlen_zero(), AST_TRANSFER_FAILED, do_proxy_auth(), get_header(), LOG_NOTICE, LOG_WARNING, and pvt_set_needdestroy().

Referenced by handle_response().

{
   enum ast_control_transfer message = AST_TRANSFER_FAILED;

   /* If no refer structure exists, then do nothing */
   if (!p->refer)
      return;

   switch (resp) {
   case 202:   /* Transfer accepted */
      /* We need  to do something here */
      /* The transferee is now sending INVITE to target */
      p->refer->status = REFER_ACCEPTED;
      /* Now wait for next message */
      ast_debug(3, "Got 202 accepted on transfer\n");
      /* We should hang along, waiting for NOTIFY's here */
      break;

   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate REFER to %s but we have no matching peer or realm auth!\n",
            ast_sockaddr_stringify(&p->recv));
         if (p->owner) {
            ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
         }
         pvt_set_needdestroy(p, "unable to authenticate REFER");
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_REFER, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", get_header(&p->initreq, "From"));
         p->refer->status = REFER_NOAUTH;
         if (p->owner) {
            ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
         }
         pvt_set_needdestroy(p, "failed to authenticate REFER");
      }
      break;
   
   case 405:   /* Method not allowed */
      /* Return to the current call onhold */
      /* Status flag needed to be reset */
      ast_log(LOG_NOTICE, "SIP transfer to %s failed, REFER not allowed. \n", p->refer->refer_to);
      pvt_set_needdestroy(p, "received 405 response");
      p->refer->status = REFER_FAILED;
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;

   case 481: /* Call leg does not exist */

      /* A transfer with Replaces did not work */
      /* OEJ: We should Set flag, cancel the REFER, go back
      to original call - but right now we can't */
      ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      pvt_set_needdestroy(p, "received 481 response");
      break;

   case 500:   /* Server error */
   case 501:   /* Method not implemented */
      /* Return to the current call onhold */
      /* Status flag needed to be reset */
      ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to);
      pvt_set_needdestroy(p, "received 500/501 response");
      p->refer->status = REFER_FAILED;
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   case 603:   /* Transfer declined */
      ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to);
      p->refer->status = REFER_FAILED;
      pvt_set_needdestroy(p, "received 603 response");
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   default:
      /* We should treat unrecognized 9xx as 900.  400 is actually
         specified as a possible response, but any 4-6xx is 
         theoretically possible. */

      if (resp < 299) { /* 1xx cases don't get here */
         ast_log(LOG_WARNING, "SIP transfer to %s had unxpected 2xx response (%d), confusion is possible. \n", p->refer->refer_to, resp);
      } else {
         ast_log(LOG_WARNING, "SIP transfer to %s with response (%d). \n", p->refer->refer_to, resp);
      }

      p->refer->status = REFER_FAILED;
      pvt_set_needdestroy(p, "received failure response");
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   }
}

static int handle_response_register	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle responses on REGISTER to services.

Definition at line 19332 of file chan_sip.c.

References __get_header(), ast_debug, ast_log(), AST_SCHED_DEL_UNREF, AST_SCHED_REPLACE_UNREF, ast_strlen_zero(), ast_tvnow(), do_register_auth(), EVENT_FLAG_SYSTEM, get_header(), LOG_NOTICE, LOG_WARNING, manager_event, MAX, pvt_set_needdestroy(), REG_STATE_NOAUTH, REG_STATE_REGISTERED, REG_STATE_REJECTED, REG_STATE_UNREGISTERED, registry_addref(), registry_unref(), regstate2str(), S_OR, sip_reregister(), sip_scheddestroy(), strcasestr(), and transmit_register().

Referenced by handle_response().

{
   int expires, expires_ms;
   struct sip_registry *r;
   r=p->registry;
   
   switch (resp) {
   case 401:   /* Unauthorized */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
         pvt_set_needdestroy(p, "failed to authenticate REGISTER");
      }
      break;
   case 403:   /* Forbidden */
      ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname);
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 403"));
      r->regstate = REG_STATE_NOAUTH;
      pvt_set_needdestroy(p, "received 403 response");
      break;
   case 404:   /* Not found */
      ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username, p->registry->hostname);
      pvt_set_needdestroy(p, "received 404 response");
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 404");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 404"));
      break;
   case 407:   /* Proxy auth */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries);
         pvt_set_needdestroy(p, "failed to authenticate REGISTER");
      }
      break;
   case 408:   /* Request timeout */
      /* Got a timeout response, so reset the counter of failed responses */
      if (r) {
         r->regattempts = 0;
      } else {
         ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid);
      }
      break;
   case 423:   /* Interval too brief */
      r->expiry = atoi(get_header(req, "Min-Expires"));
      ast_log(LOG_WARNING, "Got 423 Interval too brief for service %s@%s, minimum is %d seconds\n", p->registry->username, p->registry->hostname, r->expiry);
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 423"));
      if (r->call) {
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 423");
         pvt_set_needdestroy(p, "received 423 response");
      }
      if (r->expiry > max_expiry) {
         ast_log(LOG_WARNING, "Required expiration time from %s@%s is too high, giving up\n", p->registry->username, p->registry->hostname);
         r->expiry = r->configured_expiry;
         r->regstate = REG_STATE_REJECTED;
      } else {
         r->regstate = REG_STATE_UNREGISTERED;
         transmit_register(r, SIP_REGISTER, NULL, NULL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
      break;
   case 479:   /* SER: Not able to process the URI - address is wrong in register*/
      ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username, p->registry->hostname);
      pvt_set_needdestroy(p, "received 479 response");
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 479");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 479"));
      break;
   case 200:   /* 200 OK */
      if (!r) {
         ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
         pvt_set_needdestroy(p, "received erroneous 200 response");
         return 0;
      }
      
      r->regstate = REG_STATE_REGISTERED;
      r->regtime = ast_tvnow();     /* Reset time of last successful registration */
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
      r->regattempts = 0;
      ast_debug(1, "Registration successful\n");
      if (r->timeout > -1) {
         ast_debug(1, "Cancelling timeout %d\n", r->timeout);
      }
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 200"));
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 200");
      p->registry = registry_unref(p->registry, "unref registry entry p->registry");
      /* Let this one hang around until we have all the responses */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      /* p->needdestroy = 1; */
      
      /* set us up for re-registering
       * figure out how long we got registered for
       * according to section 6.13 of RFC, contact headers override
       * expires headers, so check those first */
      expires = 0;

      /* XXX todo: try to save the extra call */
      if (!ast_strlen_zero(get_header(req, "Contact"))) {
         const char *contact = NULL;
         const char *tmptmp = NULL;
         int start = 0;
         for(;;) {
            contact = __get_header(req, "Contact", &start);
            /* this loop ensures we get a contact header about our register request */
            if(!ast_strlen_zero(contact)) {
               if( (tmptmp=strstr(contact, p->our_contact))) {
                  contact=tmptmp;
                  break;
               }
            } else
               break;
         }
         tmptmp = strcasestr(contact, "expires=");
         if (tmptmp) {
            if (sscanf(tmptmp + 8, "%30d;", &expires) != 1)
               expires = 0;
         }
         
      }
      if (!expires)
         expires=atoi(get_header(req, "expires"));
      if (!expires)
         expires=default_expiry;
      
      expires_ms = expires * 1000;
      if (expires <= EXPIRY_GUARD_LIMIT)
         expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
      else
         expires_ms -= EXPIRY_GUARD_SECS * 1000;
      if (sipdebug)
         ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000);
      
      r->refresh= (int) expires_ms / 1000;
      
      /* Schedule re-registration before we expire */
      AST_SCHED_REPLACE_UNREF(r->expire, sched, expires_ms, sip_reregister, r,
                        registry_unref(_data,"unref in REPLACE del fail"),
                        registry_unref(r,"unref in REPLACE add fail"),
                        registry_addref(r,"The Addition side of REPLACE"));
   }
   return 1;
}

static void handle_response_subscribe	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Definition at line 19170 of file chan_sip.c.

References ast_debug, ast_free, ast_log(), ast_sched_add(), ast_string_field_set, ASTOBJ_REF, ASTOBJ_UNREF, do_proxy_auth(), get_header(), LOG_NOTICE, LOG_WARNING, pvt_set_needdestroy(), set_pvt_allowed_methods(), sip_alreadygone(), sip_subscribe_mwi_destroy(), sip_subscribe_mwi_do(), and transmit_response_with_date().

Referenced by handle_response().

{
   if (!p->mwi) {
      return;
   }

   switch (resp) {
   case 200: /* Subscription accepted */
      ast_debug(3, "Got 200 OK on subscription for MWI\n");
      set_pvt_allowed_methods(p, req);
      if (p->options) {
         ast_free(p->options);
         p->options = NULL;
      }
      p->mwi->subscribed = 1;
      if ((p->mwi->resub = ast_sched_add(sched, mwi_expiry * 1000, sip_subscribe_mwi_do, ASTOBJ_REF(p->mwi))) < 0) {
         ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      }
      break;
   case 401:
   case 407:
      ast_string_field_set(p, theirtag, NULL);
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_SUBSCRIBE, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on SUBSCRIBE to '%s'\n", get_header(&p->initreq, "From"));
         p->mwi->call = NULL;
         ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
         pvt_set_needdestroy(p, "failed to authenticate SUBSCRIBE");
      }
      break;
   case 403:
      transmit_response_with_date(p, "200 OK", req);
      ast_log(LOG_WARNING, "Authentication failed while trying to subscribe for MWI.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 403 response");
      sip_alreadygone(p);
      break;
   case 404:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that a mailbox may not have been configured.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 404 response");
      break;
   case 481:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that our dialog did not exist.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 481 response");
      break;
   case 500:
   case 501:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side may have suffered a heart attack.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 500/501 response");
      break;
   }
}

static void handle_response_update	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		int	seqno
	)			[static]

Handle authentication challenge for SIP UPDATE.

This function is only called upon the receipt of a 401/407 response to an UPDATE.

Definition at line 18592 of file chan_sip.c.

References ast_log(), do_proxy_auth(), get_header(), and LOG_NOTICE.

Referenced by handle_response().

{
   if (p->options) {
      p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
   }
   if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_UPDATE, 1)) {
      ast_log(LOG_NOTICE, "Failed to authenticate on UPDATE to '%s'\n", get_header(&p->initreq, "From"));
   }
}

static int handle_sip_publish_initial	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const int	expires
	)			[static]

Definition at line 22707 of file chan_sip.c.

References ao2_ref, event_state_compositor::callbacks, create_esc_entry(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = create_esc_entry(esc, req, expires);
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "503 Internal Server Failure", req);
      return -1;
   }

   if (esc->callbacks->initial_handler) {
      res = esc->callbacks->initial_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 0);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_modify	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag,
		const int	expires
	)			[static]

Definition at line 22757 of file chan_sip.c.

References ao2_ref, AST_SCHED_REPLACE_UNREF, event_state_compositor::callbacks, get_esc_entry(), publish_expire(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int expires_ms = expires * 1000;
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
         ao2_ref(_data, -1),
         ao2_ref(esc_entry, -1),
         ao2_ref(esc_entry, +1));

   if (esc->callbacks->modify_handler) {
      res = esc->callbacks->modify_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_refresh	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag,
		const int	expires
	)			[static]

Definition at line 22729 of file chan_sip.c.

References ao2_ref, AST_SCHED_REPLACE_UNREF, event_state_compositor::callbacks, get_esc_entry(), publish_expire(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int expires_ms = expires * 1000;
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
         ao2_ref(_data, -1),
         ao2_ref(esc_entry, -1),
         ao2_ref(esc_entry, +1));

   if (esc->callbacks->refresh_handler) {
      res = esc->callbacks->refresh_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_remove	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag
	)			[static]

Definition at line 22785 of file chan_sip.c.

References ao2_ref, ao2_unlink, AST_SCHED_DEL, event_state_compositor::callbacks, event_state_compositor::compositor, get_esc_entry(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_DEL(sched, esc_entry->sched_id);
   /* Scheduler's ref of the esc_entry */
   ao2_ref(esc_entry, -1);

   if (esc->callbacks->remove_handler) {
      res = esc->callbacks->remove_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   /* Ref from finding the esc_entry earlier in function */
   ao2_unlink(esc->compositor, esc_entry);
   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_t38_options	(	struct ast_flags *	flags,
		struct ast_flags *	mask,
		struct ast_variable *	v,
		int *	maxdatagram
	)			[static]

Handle T.38 configuration options common to users and peers.

Returns:

non-zero if any config options were handled, zero otherwise

Definition at line 25091 of file chan_sip.c.

References ast_clear_flag, ast_log(), ast_set2_flag, ast_set_flag, ast_strdupa, ast_true(), ast_variable::lineno, LOG_WARNING, ast_variable::name, strsep(), ast_variable::value, and word.

Referenced by build_peer(), and reload_config().

{
   int res = 1;

   if (!strcasecmp(v->name, "t38pt_udptl")) {
      char *buf = ast_strdupa(v->value);
      char *word, *next = buf;

      ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT);

      while ((word = strsep(&next, ","))) {
         if (ast_true(word) || !strcasecmp(word, "fec")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_FEC);
         } else if (!strcasecmp(word, "redundancy")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY);
         } else if (!strcasecmp(word, "none")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
         } else if (!strncasecmp(word, "maxdatagram=", 12)) {
            if (sscanf(&word[12], "%30u", maxdatagram) != 1) {
               ast_log(LOG_WARNING, "Invalid maxdatagram '%s' at line %d of %s\n", v->value, v->lineno, config);
               *maxdatagram = global_t38_maxdatagram;
            }
         }
      }
   } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
      ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
   } else {
      res = 0;
   }

   return res;
}

const char* hangup_cause2sip

(

int

cause

)

Convert Asterisk hangup causes to SIP codes.

 Possible values from causes.h
        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED

	In addition to these, a lot of PRI codes is defined in causes.h
	...should we take care of them too ?

	Quote RFC 3398

   ISUP Cause value                        SIP response
   ----------------                        ------------
   1  unallocated number                   404 Not Found
   2  no route to network                  404 Not found
   3  no route to destination              404 Not found
   16 normal call clearing                 --- (*)
   17 user busy                            486 Busy here
   18 no user responding                   408 Request Timeout
   19 no answer from the user              480 Temporarily unavailable
   20 subscriber absent                    480 Temporarily unavailable
   21 call rejected                        403 Forbidden (+)
   22 number changed (w/o diagnostic)      410 Gone
   22 number changed (w/ diagnostic)       301 Moved Permanently
   23 redirection to new destination       410 Gone
   26 non-selected user clearing           404 Not Found (=)
   27 destination out of order             502 Bad Gateway
   28 address incomplete                   484 Address incomplete
   29 facility rejected                    501 Not implemented
   31 normal unspecified                   480 Temporarily unavailable

Definition at line 5692 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CHAN_NOT_IMPLEMENTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_ROUTE_TRANSIT_NET, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL_UNSPECIFIED, AST_CAUSE_NOTDEFINED, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNALLOCATED, AST_CAUSE_UNREGISTERED, AST_CAUSE_USER_BUSY, and ast_debug.

Referenced by sip_hangup().

{
   switch (cause) {
      case AST_CAUSE_UNALLOCATED:      /* 1 */
      case AST_CAUSE_NO_ROUTE_DESTINATION:   /* 3 IAX2: Can't find extension in context */
      case AST_CAUSE_NO_ROUTE_TRANSIT_NET:   /* 2 */
         return "404 Not Found";
      case AST_CAUSE_CONGESTION:    /* 34 */
      case AST_CAUSE_SWITCH_CONGESTION:   /* 42 */
         return "503 Service Unavailable";
      case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
         return "408 Request Timeout";
      case AST_CAUSE_NO_ANSWER:     /* 19 */
      case AST_CAUSE_UNREGISTERED:        /* 20 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_CALL_REJECTED:    /* 21 */
         return "403 Forbidden";
      case AST_CAUSE_NUMBER_CHANGED:      /* 22 */
         return "410 Gone";
      case AST_CAUSE_NORMAL_UNSPECIFIED:  /* 31 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_INVALID_NUMBER_FORMAT:
         return "484 Address incomplete";
      case AST_CAUSE_USER_BUSY:
         return "486 Busy here";
      case AST_CAUSE_FAILURE:
         return "500 Server internal failure";
      case AST_CAUSE_FACILITY_REJECTED:   /* 29 */
         return "501 Not Implemented";
      case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
         return "503 Service Unavailable";
      /* Used in chan_iax2 */
      case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
         return "502 Bad Gateway";
      case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
         return "488 Not Acceptable Here";
         
      case AST_CAUSE_NOTDEFINED:
      default:
         ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
         return NULL;
   }

   /* Never reached */
   return 0;
}

int hangup_sip2cause

(

int

cause

)

Convert SIP hangup causes to Asterisk hangup causes.

Definition at line 5570 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_BUSY, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL, AST_CAUSE_NORMAL_TEMPORARY_FAILURE, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, AST_CAUSE_UNALLOCATED, and AST_CAUSE_USER_BUSY.

Referenced by __transmit_response(), and handle_response().

{
   /* Possible values taken from causes.h */

   switch(cause) {
      case 401:   /* Unauthorized */
         return AST_CAUSE_CALL_REJECTED;
      case 403:   /* Not found */
         return AST_CAUSE_CALL_REJECTED;
      case 404:   /* Not found */
         return AST_CAUSE_UNALLOCATED;
      case 405:   /* Method not allowed */
         return AST_CAUSE_INTERWORKING;
      case 407:   /* Proxy authentication required */
         return AST_CAUSE_CALL_REJECTED;
      case 408:   /* No reaction */
         return AST_CAUSE_NO_USER_RESPONSE;
      case 409:   /* Conflict */
         return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
      case 410:   /* Gone */
         return AST_CAUSE_NUMBER_CHANGED;
      case 411:   /* Length required */
         return AST_CAUSE_INTERWORKING;
      case 413:   /* Request entity too large */
         return AST_CAUSE_INTERWORKING;
      case 414:   /* Request URI too large */
         return AST_CAUSE_INTERWORKING;
      case 415:   /* Unsupported media type */
         return AST_CAUSE_INTERWORKING;
      case 420:   /* Bad extension */
         return AST_CAUSE_NO_ROUTE_DESTINATION;
      case 480:   /* No answer */
         return AST_CAUSE_NO_ANSWER;
      case 481:   /* No answer */
         return AST_CAUSE_INTERWORKING;
      case 482:   /* Loop detected */
         return AST_CAUSE_INTERWORKING;
      case 483:   /* Too many hops */
         return AST_CAUSE_NO_ANSWER;
      case 484:   /* Address incomplete */
         return AST_CAUSE_INVALID_NUMBER_FORMAT;
      case 485:   /* Ambiguous */
         return AST_CAUSE_UNALLOCATED;
      case 486:   /* Busy everywhere */
         return AST_CAUSE_BUSY;
      case 487:   /* Request terminated */
         return AST_CAUSE_INTERWORKING;
      case 488:   /* No codecs approved */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      case 491:   /* Request pending */
         return AST_CAUSE_INTERWORKING;
      case 493:   /* Undecipherable */
         return AST_CAUSE_INTERWORKING;
      case 500:   /* Server internal failure */
         return AST_CAUSE_FAILURE;
      case 501:   /* Call rejected */
         return AST_CAUSE_FACILITY_REJECTED;
      case 502:
         return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
      case 503:   /* Service unavailable */
         return AST_CAUSE_CONGESTION;
      case 504:   /* Gateway timeout */
         return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
      case 505:   /* SIP version not supported */
         return AST_CAUSE_INTERWORKING;
      case 600:   /* Busy everywhere */
         return AST_CAUSE_USER_BUSY;
      case 603:   /* Decline */
         return AST_CAUSE_CALL_REJECTED;
      case 604:   /* Does not exist anywhere */
         return AST_CAUSE_UNALLOCATED;
      case 606:   /* Not acceptable */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      default:
         if (cause < 500 && cause >= 400) {
            /* 4xx class error that is unknown - someting wrong with our request */
            return AST_CAUSE_INTERWORKING;
         } else if (cause < 600 && cause >= 500) {
            /* 5xx class error - problem in the remote end */
            return AST_CAUSE_CONGESTION;
         } else if (cause < 700 && cause >= 600) {
            /* 6xx - global errors in the 4xx class */
            return AST_CAUSE_INTERWORKING;
         }
         return AST_CAUSE_NORMAL;
   }
   /* Never reached */
   return 0;
}

static int init_req	(	struct sip_request *	req,
		int	sipmethod,
		const char *	recip
	)			[static]

Initialize SIP request.

Definition at line 9347 of file chan_sip.c.

References ast_free, ast_str_create(), ast_str_set(), ast_str_strlen(), sip_methods, and cfsip_methods::text.

{
   /* Initialize a request */
   memset(req, 0, sizeof(*req));
   if (!(req->data = ast_str_create(SIP_MIN_PACKET)))
      goto e_return;
   if (!(req->content = ast_str_create(SIP_MIN_PACKET)))
      goto e_free_data;
   req->method = sipmethod;
   req->header[0] = 0;
   ast_str_set(&req->data, 0, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
   req->len = ast_str_strlen(req->data);
   req->headers++;
   return 0;

e_free_data:
   ast_free(req->data);
   req->data = NULL;
e_return:
   return -1;
}

static int init_resp	(	struct sip_request *	resp,
		const char *	msg
	)			[static]

Initialize SIP response, based on SIP request.

Definition at line 9324 of file chan_sip.c.

References ast_free, ast_str_create(), and ast_str_set().

{
   /* Initialize a response */
   memset(resp, 0, sizeof(*resp));
   resp->method = SIP_RESPONSE;
   if (!(resp->data = ast_str_create(SIP_MIN_PACKET)))
      goto e_return;
   if (!(resp->content = ast_str_create(SIP_MIN_PACKET)))
      goto e_free_data;
   resp->header[0] = 0;
   ast_str_set(&resp->data, 0, "SIP/2.0 %s\r\n", msg);
   resp->len = resp->data->used;
   resp->headers++;
   return 0;

e_free_data:
   ast_free(resp->data);
   resp->data = NULL;
e_return:
   return -1;
}

static int initialize_escs

(

void

)

[static]

Definition at line 1048 of file chan_sip.c.

References ao2_container_alloc, ARRAY_LEN, esc_cmp_fn(), esc_hash_fn(), and event_state_compositors.

Referenced by load_module().

{
   int i, res = 0;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      if (!((event_state_compositors[i].compositor) =
               ao2_container_alloc(ESC_MAX_BUCKETS, esc_hash_fn, esc_cmp_fn))) {
         res = -1;
      }
   }
   return res;
}

static void initialize_initreq	(	struct sip_pvt *	p,
		struct sip_request *	req
	)			[static]

Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.

Definition at line 2815 of file chan_sip.c.

References ast_debug, ast_verbose, copy_request(), parse_request(), sip_methods, and cfsip_methods::text.

Referenced by transmit_invite(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), transmit_reinvite_with_sdp(), and update_connectedline().

{
   if (p->initreq.headers) {
      ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
   } else {
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   }
   /* Use this as the basis */
   copy_request(&p->initreq, req);
   parse_request(&p->initreq);
   if (req->debug) {
      ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
   }
}

static void initreqprep	(	struct sip_request *	req,
		struct sip_pvt *	p,
		int	sipmethod,
		const char *const	explicit_uri
	)			[static]

Initiate new SIP request to peer/user.

Todo:

Need to add back the VXML URL here at some point, possibly use build_string for all this junk

Definition at line 11026 of file chan_sip.c.

References add_header(), add_header_max_forwards(), add_route(), AST_DIGIT_ANYNUM, ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_sockaddr_port, ast_sockaddr_stringify_host(), ast_str_alloca, ast_str_append(), ast_str_set(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_encode(), build_contact(), exten, init_req(), ourport, S_OR, sip_cfg, sip_methods, sip_standard_port(), and cfsip_methods::text.

Referenced by transmit_invite(), and transmit_notify_with_mwi().

{
   struct ast_str *invite = ast_str_alloca(256);
   char from[256];
   char to[256];
   char tmp_n[SIPBUFSIZE/2];  /* build a local copy of 'n' if needed */
   char tmp_l[SIPBUFSIZE/2];  /* build a local copy of 'l' if needed */
   const char *l = NULL;   /* XXX what is this, exactly ? */
   const char *n = NULL;   /* XXX what is this, exactly ? */
   const char *d = NULL;   /* domain in from header */
   const char *urioptions = "";
   int ourport;

   if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
      const char *s = p->username;  /* being a string field, cannot be NULL */

      /* Test p->username against allowed characters in AST_DIGIT_ANY
         If it matches the allowed characters list, then sipuser = ";user=phone"
         If not, then sipuser = ""
      */
      /* + is allowed in first position in a tel: uri */
      if (*s == '+')
         s++;
      for (; *s; s++) {
         if (!strchr(AST_DIGIT_ANYNUM, *s) )
            break;
      }
      /* If we have only digits, add ;user=phone to the uri */
      if (!*s)
         urioptions = ";user=phone";
   }


   snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);

   d = S_OR(p->fromdomain, ast_sockaddr_stringify_host(&p->ourip));
   if (p->owner) {
      if ((ast_party_id_presentation(&p->owner->connected.id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
         l = p->owner->connected.id.number.valid ? p->owner->connected.id.number.str : NULL;
         n = p->owner->connected.id.name.valid ? p->owner->connected.id.name.str : NULL;
      } else if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
         /* if we are not sending RPID and user wants his callerid restricted */    
         l = CALLERID_UNKNOWN;
         n = l;
         d = FROMDOMAIN_INVALID;
      }
   }

   /* Hey, it's a NOTIFY! See if they've configured a mwi_from.
    * XXX Right now, this logic works because the only place that mwi_from
    * is set on the sip_pvt is in sip_send_mwi_to_peer. If things changed, then
    * we might end up putting the mwi_from setting into other types of NOTIFY
    * messages as well.
    */
   if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->mwi_from)) {
      l = p->mwi_from;
   }

   if (ast_strlen_zero(l))
      l = default_callerid;
   if (ast_strlen_zero(n))
      n = l;
   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromuser))
      l = p->fromuser;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromuser, l);

   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromname))
      n = p->fromname;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromname, n);

   if (sip_cfg.pedanticsipchecking) {
      ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
      n = tmp_n;
      ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0);
      l = tmp_l;
   }

   ourport = (p->fromdomainport) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
   if (!sip_standard_port(p->socket.type, ourport)) {
      snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, d, ourport, p->tag);
   } else {
      snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, d, p->tag);
   }

   if (!ast_strlen_zero(explicit_uri)) {
      ast_str_set(&invite, 0, "%s", explicit_uri);
   } else {
      /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
      if (!ast_strlen_zero(p->fullcontact)) {
         /* If we have full contact, trust it */
         ast_str_append(&invite, 0, "%s", p->fullcontact);
      } else {
         /* Otherwise, use the username while waiting for registration */
         ast_str_append(&invite, 0, "sip:");
         if (!ast_strlen_zero(p->username)) {
            n = p->username;
            if (sip_cfg.pedanticsipchecking) {
               ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
               n = tmp_n;
            }
            ast_str_append(&invite, 0, "%s@", n);
         }
         ast_str_append(&invite, 0, "%s", p->tohost);
         if (p->portinuri) {
            ast_str_append(&invite, 0, ":%d", ast_sockaddr_port(&p->sa));
         }
         ast_str_append(&invite, 0, "%s", urioptions);
      }
   }

   /* If custom URI options have been provided, append them */
   if (p->options && !ast_strlen_zero(p->options->uri_options))
      ast_str_append(&invite, 0, ";%s", p->options->uri_options);
   
   /* This is the request URI, which is the next hop of the call
      which may or may not be the destination of the call
   */
   ast_string_field_set(p, uri, invite->str);

   if (!ast_strlen_zero(p->todnid)) {
      /*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
      if (!strchr(p->todnid, '@')) {
         /* We have no domain in the dnid */
         snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      } else {
         snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      }
   } else {
      if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
         /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
         snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
      } else if (p->options && p->options->vxml_url) {
         /* If there is a VXML URL append it to the SIP URL */
         snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
      } else {
         snprintf(to, sizeof(to), "<%s>", p->uri);
      }
   }

   init_req(req, sipmethod, p->uri);
   /* now tmp_n is available so reuse it to build the CSeq */
   snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);

   add_header(req, "Via", p->via);
   add_header_max_forwards(p, req);
   /* This will be a no-op most of the time. However, under certain circumstances,
    * NOTIFY messages will use this function for preparing the request and should
    * have Route headers present.
    */
   add_route(req, p->route);

   add_header(req, "From", from);
   add_header(req, "To", to);
   ast_string_field_set(p, exten, l);
   build_contact(p);
   add_header(req, "Contact", p->our_contact);
   add_header(req, "Call-ID", p->callid);
   add_header(req, "CSeq", tmp_n);
   if (!ast_strlen_zero(global_useragent)) {
      add_header(req, "User-Agent", global_useragent);
   }
}

static const char * insecure2str

(

int

mode

)

[static]

Convert Insecure setting to printable string.

Definition at line 15775 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer().

{
   return map_x_s(insecurestr, mode, "<error>");
}

static int interpret_t38_parameters	(	struct sip_pvt *	p,
		const struct ast_control_t38_parameters *	parameters
	)			[static]

Helper function which updates T.38 capability information and triggers a reinvite.

Definition at line 6222 of file chan_sip.c.

References AST_CONTROL_T38_PARAMETERS, ast_queue_control_data(), AST_SCHED_DEL, AST_SCHED_DEL_UNREF, ast_set_flag, AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_PARMS, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_test_flag, ast_udptl_get_far_max_ifp(), ast_udptl_set_local_max_ifp(), change_t38_state(), FALSE, ast_control_t38_parameters::fill_bit_removal, ast_control_t38_parameters::max_ifp, MIN, ast_control_t38_parameters::request_response, transmit_reinvite_with_sdp(), transmit_response_reliable(), transmit_response_with_t38_sdp(), and TRUE.

Referenced by sip_indicate().

{
   int res = 0;

   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) || !p->udptl) {
      return -1;
   }
   switch (parameters->request_response) {
   case AST_T38_NEGOTIATED:
   case AST_T38_REQUEST_NEGOTIATE:         /* Request T38 */
      /* Negotiation can not take place without a valid max_ifp value. */
      if (!parameters->max_ifp) {
         change_t38_state(p, T38_DISABLED);
         if (p->t38.state == T38_PEER_REINVITE) {
            AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
            transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
         }
         break;
      } else if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
         p->t38.our_parms = *parameters;
         /* modify our parameters to conform to the peer's parameters,
          * based on the rules in the ITU T.38 recommendation
          */
         if (!p->t38.their_parms.fill_bit_removal) {
            p->t38.our_parms.fill_bit_removal = FALSE;
         }
         if (!p->t38.their_parms.transcoding_mmr) {
            p->t38.our_parms.transcoding_mmr = FALSE;
         }
         if (!p->t38.their_parms.transcoding_jbig) {
            p->t38.our_parms.transcoding_jbig = FALSE;
         }
         p->t38.our_parms.version = MIN(p->t38.our_parms.version, p->t38.their_parms.version);
         p->t38.our_parms.rate_management = p->t38.their_parms.rate_management;
         ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
         change_t38_state(p, T38_ENABLED);
         transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
      } else if (p->t38.state != T38_ENABLED) {
         p->t38.our_parms = *parameters;
         ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
         change_t38_state(p, T38_LOCAL_REINVITE);
         if (!p->pendinginvite) {
            transmit_reinvite_with_sdp(p, TRUE, FALSE);
         } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
            ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
         }
      }
      break;
   case AST_T38_TERMINATED:
   case AST_T38_REFUSED:
   case AST_T38_REQUEST_TERMINATE:         /* Shutdown T38 */
      if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
         change_t38_state(p, T38_DISABLED);
         transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
      } else if (p->t38.state == T38_ENABLED)
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      break;
   case AST_T38_REQUEST_PARMS: {    /* Application wants remote's parameters re-sent */
      struct ast_control_t38_parameters parameters = p->t38.their_parms;

      if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL(sched, p->t38id);
         parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
         parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
         ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
         /* we need to return a positive value here, so that applications that
          * send this request can determine conclusively whether it was accepted or not...
          * older versions of chan_sip would just silently accept it and return zero.
          */
         res = AST_T38_REQUEST_PARMS;
      }
      break;
   }
   default:
      res = -1;
      break;
   }

   return res;
}

static int is_method_allowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)			[static]

Check if method is allowed for a device or a dialog.

Definition at line 7604 of file chan_sip.c.

Referenced by sip_sendtext(), and update_connectedline().

{
   return ((*allowed_methods) >> method) & 1;
}

static void list_route

(

struct sip_route *

route

)

[static]

List all routes - mostly for debugging.

Definition at line 13112 of file chan_sip.c.

References ast_verbose.

Referenced by build_route().

{
   if (!route) {
      ast_verbose("list_route: no route\n");
   } else {
      for (;route; route = route->next)
         ast_verbose("list_route: hop: <%s>\n", route->hop);
   }
}

static int load_module

(

void

)

[static]

PBX load module - initialization.

Definition at line 28364 of file chan_sip.c.

References ao2_container_alloc, ao2_t_container_alloc, ARRAY_LEN, ast_cc_agent_register(), ast_cc_monitor_register(), ast_channel_register(), ast_check_realtime(), ast_cli_register_multiple(), ast_custom_function_register, ast_data_register_multiple, ast_log(), ast_manager_register_xml, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_SUCCESS, ast_realtime_require_field(), ast_register_application_xml, ast_rtp_glue_register, AST_TEST_REGISTER, ast_udptl_proto_register(), ast_verbose, ASTOBJ_CONTAINER_INIT, CHANNEL_MODULE_LOAD, dialog_cmp_cb(), dialog_hash_cb(), EVENT_FLAG_REPORTING, EVENT_FLAG_SYSTEM, initialize_escs(), io_context_create(), io_context_destroy(), LOG_ERROR, manager_show_registry(), manager_sip_qualify_peer(), manager_sip_show_peer(), manager_sip_show_peers(), manager_sipnotify(), network_change_event_subscribe(), peer_cmp_cb(), peer_hash_cb(), peer_ipcmp_cb(), peer_iphash_cb(), regl, reload_config(), restart_monitor(), RQ_CHAR, RQ_INTEGER4, RQ_UINTEGER2, sched_context_create(), sched_context_destroy(), ast_channel_tech::send_digit_begin, SENTINEL, sip_addheader(), sip_dtmfmode(), sip_epa_register(), sip_is_xml_parsable(), sip_monitor_instance_cmp_fn(), sip_monitor_instance_hash_fn(), sip_poke_all_peers(), sip_register_tests(), sip_removeheader(), sip_send_all_mwi_subscriptions(), sip_send_all_registers(), sip_tech_info, submwil, threadt_cmp_cb(), and threadt_hash_cb().

{
   ast_verbose("SIP channel loading...\n");
   /* the fact that ao2_containers can't resize automatically is a major worry! */
   /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
   peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers");
   peers_by_ip = ao2_t_container_alloc(HASH_PEER_SIZE, peer_iphash_cb, peer_ipcmp_cb, "allocate peers_by_ip");
   dialogs = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs");
   threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table");
   
   ASTOBJ_CONTAINER_INIT(&regl); /* Registry object list -- not searched for anything */
   ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */

   if (!(sched = sched_context_create())) {
      ast_log(LOG_ERROR, "Unable to create scheduler context\n");
      return AST_MODULE_LOAD_FAILURE;
   }

   if (!(io = io_context_create())) {
      ast_log(LOG_ERROR, "Unable to create I/O context\n");
      sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }

   sip_reloadreason = CHANNEL_MODULE_LOAD;

   can_parse_xml = sip_is_xml_parsable();
   if(reload_config(sip_reloadreason)) {  /* Load the configuration from sip.conf */
      return AST_MODULE_LOAD_DECLINE;
   }

   /* Prepare the version that does not require DTMF BEGIN frames.
    * We need to use tricks such as memcpy and casts because the variable
    * has const fields.
    */
   memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
   memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));

   /* Make sure we can register our sip channel type */
   if (ast_channel_register(&sip_tech)) {
      ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
      io_context_destroy(io);
      sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }

#ifdef TEST_FRAMEWORK
   AST_TEST_REGISTER(test_sip_peers_get);
#endif

   /* Register AstData providers */
   ast_data_register_multiple(sip_data_providers, ARRAY_LEN(sip_data_providers));

   /* Register all CLI functions for SIP */
   ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));

   /* Tell the UDPTL subdriver that we're here */
   ast_udptl_proto_register(&sip_udptl);

   /* Tell the RTP engine about our RTP glue */
   ast_rtp_glue_register(&sip_rtp_glue);

   /* Register dialplan applications */
   ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
   ast_register_application_xml(app_sipaddheader, sip_addheader);
   ast_register_application_xml(app_sipremoveheader, sip_removeheader);

   /* Register dialplan functions */
   ast_custom_function_register(&sip_header_function);
   ast_custom_function_register(&sippeer_function);
   ast_custom_function_register(&sipchaninfo_function);
   ast_custom_function_register(&checksipdomain_function);

   /* Register manager commands */
   ast_manager_register_xml("SIPpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peers);
   ast_manager_register_xml("SIPshowpeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peer);
   ast_manager_register_xml("SIPqualifypeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_qualify_peer);
   ast_manager_register_xml("SIPshowregistry", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_show_registry);
   ast_manager_register_xml("SIPnotify", EVENT_FLAG_SYSTEM, manager_sipnotify);
   sip_poke_all_peers();   
   sip_send_all_registers();
   sip_send_all_mwi_subscriptions();
   initialize_escs();

   if (sip_epa_register(&cc_epa_static_data)) {
      return AST_MODULE_LOAD_DECLINE;
   }

   if (sip_reqresp_parser_init() == -1) {
      ast_log(LOG_ERROR, "Unable to initialize the SIP request and response parser\n");
      return AST_MODULE_LOAD_DECLINE;
   }

   if (can_parse_xml) {
      /* SIP CC agents require the ability to parse XML PIDF bodies
       * in incoming PUBLISH requests
       */
      if (ast_cc_agent_register(&sip_cc_agent_callbacks)) {
         return AST_MODULE_LOAD_DECLINE;
      }
   }
   if (ast_cc_monitor_register(&sip_cc_monitor_callbacks)) {
      return AST_MODULE_LOAD_DECLINE;
   }
   if (!(sip_monitor_instances = ao2_container_alloc(37, sip_monitor_instance_hash_fn, sip_monitor_instance_cmp_fn))) {
      return AST_MODULE_LOAD_DECLINE;
   }

   /* And start the monitor for the first time */
   restart_monitor();

   ast_realtime_require_field(ast_check_realtime("sipregs") ? "sipregs" : "sippeers",
      "name", RQ_CHAR, 10,
      "ipaddr", RQ_CHAR, 15,
      "port", RQ_UINTEGER2, 5,
      "regseconds", RQ_INTEGER4, 11,
      "defaultuser", RQ_CHAR, 10,
      "fullcontact", RQ_CHAR, 35,
      "regserver", RQ_CHAR, 20,
      "useragent", RQ_CHAR, 20,
      "lastms", RQ_INTEGER4, 11,
      SENTINEL);


   sip_register_tests();
   network_change_event_subscribe();

   return AST_MODULE_LOAD_SUCCESS;
}

static int local_attended_transfer	(	struct sip_pvt *	transferer,
		struct sip_dual *	current,
		struct sip_request *	req,
		int	seqno,
		int *	nounlock
	)			[static]

Find all call legs and bridge transferee with target called from handle_request_refer.

Note:

this function assumes two locks to begin with, sip_pvt transferer and current.chan1 (the pvt's owner)... 2 additional locks are held at the beginning of the function, targetcall_pvt, and targetcall_pvt's owner channel (which is stored in target.chan1). These 2 locks _MUST_ be let go by the end of the function. Do not be confused into thinking a pvt's owner is the same thing as the channels locked at the beginning of this function, after the masquerade this may not be true. Be consistent and unlock only the exact same pointers that were locked to begin with.

If this function is successful, only the transferer pvt lock will remain on return. Setting nounlock indicates to handle_request_do() that the pvt's owner it locked does not require an unlock.

Definition at line 21640 of file chan_sip.c.

References ao2_t_ref, append_history, ast_bridged_channel(), AST_CEL_ATTENDEDTRANSFER, ast_cel_report_event(), ast_channel_queue_connected_line_update(), ast_channel_ref, ast_channel_unlock, ast_channel_unref, ast_clear_flag, ast_connected_line_build_data(), AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, AST_CONTROL_READ_ACTION, AST_CONTROL_UNHOLD, ast_copy_string(), ast_debug, ast_do_masquerade(), AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO, ast_indicate(), ast_manager_event_multichan, ast_party_connected_line_copy(), ast_party_connected_line_free(), ast_party_connected_line_init(), ast_queue_control_data(), ast_set_flag, ast_state2str(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_streamfile(), ast_strlen_zero(), ast_waitstream(), attempt_transfer(), EVENT_FLAG_CALL, frame_size, get_sip_pvt_byid_locked(), ast_channel::name, pbx_builtin_getvar_helper(), sip_pvt_lock, sip_pvt_unlock, ast_party_connected_line::source, transmit_notify_with_sipfrag(), transmit_response(), TRUE, and xfersound.

Referenced by handle_request_refer().

{
   struct sip_dual target;    /* Chan 1: Call from tranferer to Asterisk */
               /* Chan 2: Call from Asterisk to target */
   int res = 0;
   struct sip_pvt *targetcall_pvt;
   struct ast_party_connected_line connected_to_transferee;
   struct ast_party_connected_line connected_to_target;
   char transferer_linkedid[32];
   struct ast_channel *chans[2];

   /* Check if the call ID of the replaces header does exist locally */
   if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag,
      transferer->refer->replaces_callid_fromtag))) {
      if (transferer->refer->localtransfer) {
         /* We did not find the refered call. Sorry, can't accept then */
         transmit_response(transferer, "202 Accepted", req);
         /* Let's fake a response from someone else in order
            to follow the standard */
         transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE);
         append_history(transferer, "Xfer", "Refer failed");
         ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
         transferer->refer->status = REFER_FAILED;
         return -1;
      }
      /* Fall through for remote transfers that we did not find locally */
      ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n");
      return 0;
   }

   /* Ok, we can accept this transfer */
   transmit_response(transferer, "202 Accepted", req);
   append_history(transferer, "Xfer", "Refer accepted");
   if (!targetcall_pvt->owner) { /* No active channel */
      ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
      /* Cancel transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
      transferer->refer->status = REFER_FAILED;
      sip_pvt_unlock(targetcall_pvt);
      if (targetcall_pvt)
         ao2_t_ref(targetcall_pvt, -1, "Drop targetcall_pvt pointer");
      return -1;
   }

   /* We have a channel, find the bridge */
   target.chan1 = targetcall_pvt->owner;           /* Transferer to Asterisk */
   target.chan2 = ast_bridged_channel(targetcall_pvt->owner);  /* Asterisk to target */

   if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) {
      /* Wrong state of new channel */
      if (target.chan2)
         ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
      else if (target.chan1->_state != AST_STATE_RING)
         ast_debug(4, "SIP attended transfer: Error: No target channel\n");
      else
         ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n");
   }

   /* Transfer */
   if (sipdebug) {
      if (current->chan2)  /* We have two bridges */
         ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name);
      else        /* One bridge, propably transfer of IVR/voicemail etc */
         ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name);
   }

   ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */

   ast_copy_string(transferer_linkedid, transferer->owner->linkedid, sizeof(transferer_linkedid));

   /* Perform the transfer */
   chans[0] = transferer->owner;
   chans[1] = target.chan1;
   ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans, "TransferMethod: SIP\r\nTransferType: Attended\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\n",
      transferer->owner->name,
      transferer->owner->uniqueid,
      transferer->callid,
      target.chan1->name,
      target.chan1->uniqueid);
   ast_party_connected_line_init(&connected_to_transferee);
   ast_party_connected_line_init(&connected_to_target);
   /* No need to lock current->chan1 here since it was locked in sipsock_read */
   ast_party_connected_line_copy(&connected_to_transferee, &current->chan1->connected);
   /* No need to lock target.chan1 here since it was locked in get_sip_pvt_byid_locked */
   ast_party_connected_line_copy(&connected_to_target, &target.chan1->connected);
   connected_to_target.source = connected_to_transferee.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
   res = attempt_transfer(current, &target);
   if (res) {
      /* Failed transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
      /* if transfer failed, go ahead and unlock targetcall_pvt and it's owner channel */
      sip_pvt_unlock(targetcall_pvt);
      ast_channel_unlock(target.chan1);
   } else {
      /* Transfer succeeded! */
      const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");

      /* target.chan1 was locked in get_sip_pvt_byid_locked, do not unlock target.chan1 before this */
      ast_cel_report_event(target.chan1, AST_CEL_ATTENDEDTRANSFER, NULL, transferer_linkedid, target.chan2);

      /* Tell transferer that we're done. */
      transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
      append_history(transferer, "Xfer", "Refer succeeded");
      transferer->refer->status = REFER_200OK;
      if (target.chan2 && !ast_strlen_zero(xfersound) && ast_streamfile(target.chan2, xfersound, target.chan2->language) >= 0) {
         ast_waitstream(target.chan2, "");
      }

      /* By forcing the masquerade, we know that target.chan1 and target.chan2 are bridged. We then
       * can queue connected line updates where they need to go.
       *
       * before a masquerade, all channel and pvt locks must be unlocked.  Any recursive
       * channel locks held before this function invalidates channel container locking order.
       * Since we are unlocking both the pvt (transferer) and its owner channel (current.chan1)
       * it is possible for current.chan1 to be destroyed in the pbx thread.  To prevent this
       * we must give c a reference before any unlocking takes place.
       */

      ast_channel_ref(current->chan1);
      ast_channel_unlock(current->chan1); /* current.chan1 is p->owner before the masq, it was locked by socket_read()*/
      ast_channel_unlock(target.chan1);
      *nounlock = 1;  /* we just unlocked the dialog's channel and have no plans of locking it again. */
      sip_pvt_unlock(targetcall_pvt);
      sip_pvt_unlock(transferer);

      ast_do_masquerade(target.chan1);

      sip_pvt_lock(transferer); /* the transferer pvt is expected to remain locked on return */

      ast_indicate(target.chan1, AST_CONTROL_UNHOLD);

      if (target.chan2) {
         ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
         ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);
      } else {
         /* Since target.chan1 isn't actually connected to another channel, there is no way for us
          * to queue a frame so that its connected line status will be updated.
          *
          * Instead, we use the somewhat hackish approach of using a special control frame type that
          * instructs ast_read to perform a specific action. In this case, the frame we queue tells
          * ast_read to call the connected line interception macro configured for target.chan1.
          */
         struct ast_control_read_action_payload *frame_payload;
         int payload_size;
         int frame_size;
         unsigned char connected_line_data[1024];
         payload_size = ast_connected_line_build_data(connected_line_data,
            sizeof(connected_line_data), &connected_to_target, NULL);
         frame_size = payload_size + sizeof(*frame_payload);
         if (payload_size != -1 && (frame_payload = alloca(frame_size))) {
            frame_payload->payload_size = payload_size;
            memcpy(frame_payload->payload, connected_line_data, payload_size);
            frame_payload->action = AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO;
            ast_queue_control_data(target.chan1, AST_CONTROL_READ_ACTION, frame_payload, frame_size);
         }
         /* In addition to queueing the read action frame so that target.chan1's connected line info
          * will be updated, we also are going to queue a plain old connected line update on target.chan1. This
          * way, either Dial or Queue can apply this connected line update to the outgoing ringing channel.
          */
         ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);

      }
      ast_channel_unref(current->chan1);
   }

   /* at this point if the transfer is successful only the transferer pvt should be locked. */
   ast_party_connected_line_free(&connected_to_target);
   ast_party_connected_line_free(&connected_to_transferee);
   if (targetcall_pvt)
      ao2_t_ref(targetcall_pvt, -1, "drop targetcall_pvt");
   return 1;
}

static int lws2sws	(	char *	msgbuf,
		int	len
	)			[static]

Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.

Definition at line 7692 of file chan_sip.c.

Referenced by handle_request_do().

{
   int h = 0, t = 0;
   int lws = 0;

   for (; h < len;) {
      /* Eliminate all CRs */
      if (msgbuf[h] == '\r') {
         h++;
         continue;
      }
      /* Check for end-of-line */
      if (msgbuf[h] == '\n') {
         /* Check for end-of-message */
         if (h + 1 == len)
            break;
         /* Check for a continuation line */
         if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
            /* Merge continuation line */
            h++;
            continue;
         }
         /* Propagate LF and start new line */
         msgbuf[t++] = msgbuf[h++];
         lws = 0;
         continue;
      }
      if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
         if (lws) {
            h++;
            continue;
         }
         msgbuf[t++] = msgbuf[h++];
         lws = 1;
         continue;
      }
      msgbuf[t++] = msgbuf[h++];
      if (lws)
         lws = 0;
   }
   msgbuf[t] = '\0';
   return t;
}

static void make_our_tag	(	char *	tagbuf,
		size_t	len
	)			[static]

Make our SIP dialog tag.

Definition at line 7004 of file chan_sip.c.

References ast_random().

Referenced by handle_request_invite(), handle_request_subscribe(), sip_alloc(), transmit_register(), and transmit_response_using_temp().

{
   snprintf(tagbuf, len, "as%08lx", ast_random());
}

static int manager_show_registry	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP registrations in the manager API.

Definition at line 15413 of file chan_sip.c.

References ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, regl, regstate2str(), S_OR, and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Registrations will follow", "start");

   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_RDLOCK(iterator);
      astman_append(s,
         "Event: RegistryEntry\r\n"
         "%s"
         "Host: %s\r\n"
         "Port: %d\r\n"
         "Username: %s\r\n"
         "Domain: %s\r\n"
         "DomainPort: %d\r\n"
         "Refresh: %d\r\n"
         "State: %s\r\n"
         "RegistrationTime: %ld\r\n"
         "\r\n",
         idtext,
         iterator->hostname,
         iterator->portno ? iterator->portno : STANDARD_SIP_PORT,
         iterator->username,
         S_OR(iterator->regdomain,iterator->hostname),
         iterator->regdomainport ? iterator->regdomainport : STANDARD_SIP_PORT,
         iterator->refresh,
         regstate2str(iterator->regstate),
         (long) iterator->regtime.tv_sec);
      ASTOBJ_UNLOCK(iterator);
      total++;
   } while(0));

   astman_append(s,
      "Event: RegistrationsComplete\r\n"
      "EventList: Complete\r\n"
      "ListItems: %d\r\n"
      "%s"
      "\r\n", total, idtext);
   
   return 0;
}

static int manager_sip_qualify_peer	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Qualify SIP peers in the manager API.

Definition at line 16133 of file chan_sip.c.

References _sip_qualify_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), and astman_send_error().

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "qualify";
   a[2] = "peer";
   a[3] = peer;

   _sip_qualify_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n\r\n" );
   return 0;
}

static int manager_sip_show_peer	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP peers in the manager API.

Definition at line 16074 of file chan_sip.c.

References _sip_show_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), and astman_send_error().

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "show";
   a[2] = "peer";
   a[3] = peer;

   _sip_show_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n\r\n" );
   return 0;
}

static int manager_sip_show_peers	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Show SIP peers in the manager API.

Definition at line 15463 of file chan_sip.c.

References _sip_show_peers(), ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   const char *a[] = {"sip", "show", "peers"};
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Peer status list will follow", "start");
   /* List the peers in separate manager events */
   _sip_show_peers(-1, &total, s, m, 3, a);
   /* Send final confirmation */
   astman_append(s,
   "Event: PeerlistComplete\r\n"
   "EventList: Complete\r\n"
   "ListItems: %d\r\n"
   "%s"
   "\r\n", total, idtext);
   return 0;
}

static int manager_sipnotify	(	struct mansession *	s,
		const struct message *	m
	)			[static]

Definition at line 11920 of file chan_sip.c.

References ast_log(), ast_set_flag, ast_str_append(), ast_str_strlen(), ast_strlen_zero(), ast_variable_new(), ast_variables_destroy(), astman_get_header(), astman_get_variables(), astman_send_ack(), astman_send_error(), create_addr(), dialog_unlink_all(), LOG_WARNING, ast_variable::name, ast_variable::next, sip_alloc(), sip_notify_allocate(), sip_scheddestroy(), transmit_invite(), TRUE, ast_variable::value, and var.

Referenced by load_module().

{
   const char *channame = astman_get_header(m, "Channel");
   struct ast_variable *vars = astman_get_variables(m);
   struct sip_pvt *p;
   struct ast_variable *header, *var;

   if (ast_strlen_zero(channame)) {
      astman_send_error(s, m, "SIPNotify requires a channel name");
      return 0;
   }

   if (!strncasecmp(channame, "sip/", 4)) {
      channame += 4;
   }

   if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL))) {
      astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
      return 0;
   }

   if (create_addr(p, channame, NULL, 0, NULL)) {
      /* Maybe they're not registered, etc. */
      dialog_unlink_all(p, TRUE, TRUE);
      dialog_unref(p, "unref dialog inside for loop" );
      /* sip_destroy(p); */
      astman_send_error(s, m, "Could not create address");
      return 0;
   }

   /* Notify is outgoing call */
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
   sip_notify_allocate(p);

   p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");

   for (var = vars; var; var = var->next) {
      if (!strcasecmp(var->name, "Content")) {
         if (ast_str_strlen(p->notify->content))
            ast_str_append(&p->notify->content, 0, "\r\n");
         ast_str_append(&p->notify->content, 0, "%s", var->value);
      } else if (!strcasecmp(var->name, "Content-Length")) {
         ast_log(LOG_WARNING, "it is not necessary to specify Content-Length, ignoring");
      } else {
         header->next = ast_variable_new(var->name, var->value, "");
         header = header->next;
      }
   }

   dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
   sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
   transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);

   astman_send_ack(s, m, "Notify Sent");
   ast_variables_destroy(vars);
   return 0;
}

static int map_s_x	(	const struct _map_x_s *	table,
		const char *	s,
		int	errorvalue
	)			[static]

map from a string to an integer value, case insensitive. If no match is found, return errorvalue.

Definition at line 2212 of file chan_sip.c.

Referenced by str2dtmfmode(), str2stmode(), and str2strefresher().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++)
      if (!strcasecmp(cur->s, s))
         return cur->x;
   return errorvalue;
}

static const char* map_x_s	(	const struct _map_x_s *	table,
		int	x,
		const char *	errorstring
	)			[static]

map from an integer value to a string. If no match is found, return errorstring

Definition at line 2199 of file chan_sip.c.

Referenced by dtmfmode2str(), faxec2str(), insecure2str(), referstatus2str(), regstate2str(), stmode2str(), and strefresher2str().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++)
      if (cur->x == x)
         return cur->s;
   return errorstring;
}

static void mark_method_allowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)			[static]

Definition at line 7593 of file chan_sip.c.

Referenced by handle_response(), mark_parsed_methods(), and set_pvt_allowed_methods().

{
   (*allowed_methods) |= (1 << method);
}

static void mark_method_unallowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)			[static]

Definition at line 7598 of file chan_sip.c.

Referenced by handle_response(), and handle_response_publish().

{
   (*allowed_methods) &= ~(1 << method);
}

static void mark_parsed_methods	(	unsigned int *	methods,
		char *	methods_str
	)			[static]

Definition at line 7609 of file chan_sip.c.

References ast_skip_blanks(), ast_strlen_zero(), find_sip_method(), mark_method_allowed(), and strsep().

Referenced by build_peer(), parse_allowed_methods(), and reload_config().

{
   char *method;
   for (method = strsep(&methods_str, ","); !ast_strlen_zero(method); method = strsep(&methods_str, ",")) {
      int id = find_sip_method(ast_skip_blanks(method));
      if (id == SIP_UNKNOWN) {
         continue;
      }
      mark_method_allowed(methods, id);
   }
}

static enum match_req_res match_req_to_dialog	(	struct sip_pvt *	sip_pvt_ptr,
		struct match_req_args *	arg
	)			[static]

Definition at line 7204 of file chan_sip.c.

References ast_strlen_zero(), ast_test_flag, match_req_args::authentication_present, match_req_args::callid, match_req_args::fromtag, match_req_args::method, match_req_args::ruri, match_req_args::seqno, SIP_REQ_LOOP_DETECTED, SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, match_req_args::totag, match_req_args::viabranch, and match_req_args::viasentby.

Referenced by find_call().

{
   const char *init_ruri = NULL;
   if (sip_pvt_ptr->initreq.headers) {
      init_ruri = REQ_OFFSET_TO_STR(&sip_pvt_ptr->initreq, rlPart2);
   }

   /*
    * Match Tags and call-id to Dialog
    */
   if (!ast_strlen_zero(arg->callid) && strcmp(sip_pvt_ptr->callid, arg->callid)) {
      /* call-id does not match. */
      return SIP_REQ_NOT_MATCH;
   }
   if (arg->method == SIP_RESPONSE) {
      /* Verify totag if we have one stored for this dialog, but never be strict about this for
       * a response until the dialog is established */
      if (!ast_strlen_zero(sip_pvt_ptr->theirtag) && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
         if (ast_strlen_zero(arg->totag)) {
            /* missing totag when they already gave us one earlier */
            return SIP_REQ_NOT_MATCH;
         }
         if (strcmp(arg->totag, sip_pvt_ptr->theirtag)) {
            /* The totag of the response does not match the one we have stored */
            return SIP_REQ_NOT_MATCH;
         }
      }
      /* Verify fromtag of response matches the tag we gave them. */
      if (strcmp(arg->fromtag, sip_pvt_ptr->tag)) {
         /* fromtag from response does not match our tag */
         return SIP_REQ_NOT_MATCH;
      }
   } else {
      /* Verify the fromtag of Request matches the tag they provided earlier. */
      if (strcmp(arg->fromtag, sip_pvt_ptr->theirtag)) {
         /* their tag does not match the one was have stored for them */
         return SIP_REQ_NOT_MATCH;
      }
      /* Verify if totag is present in Request, that it matches what we gave them as our tag earlier */
      if (!ast_strlen_zero(arg->totag) && (strcmp(arg->totag, sip_pvt_ptr->tag))) {
         /* totag from Request does not match our tag */
         return SIP_REQ_NOT_MATCH;
      }
   }

   /*
    * Compare incoming request against initial transaction.
    * 
    * This is a best effort attempt at distinguishing forked requests from
    * our initial transaction.  If all the elements are NOT in place to evaluate
    * this, this block is ignored and the dialog match is made regardless.
    * Once the totag is established after the dialog is confirmed, this is not necessary.
    *
    * CRITERIA required for initial transaction matching.
    * 
    * 1. Is a Request
    * 2. Callid and theirtag match (this is done in the dialog matching block)
    * 3. totag is NOT present
    * 4. CSeq matchs our initial transaction's cseq number
    * 5. pvt has init via branch parameter stored
    */
   if ((arg->method != SIP_RESPONSE) &&                 /* must be a Request */
      ast_strlen_zero(arg->totag) &&                   /* must not have a totag */
      (sip_pvt_ptr->init_icseq == arg->seqno) &&       /* the cseq must be the same as this dialogs initial cseq */
      !ast_strlen_zero(sip_pvt_ptr->initviabranch) &&  /* The dialog must have started with a RFC3261 compliant branch tag */
      init_ruri) {                                     /* the dialog must have an initial request uri associated with it */
      /* This Request matches all the criteria required for Loop/Merge detection.
       * Now we must go down the path of comparing VIA's and RURIs. */
      if (ast_strlen_zero(arg->viabranch) ||
         strcmp(arg->viabranch, sip_pvt_ptr->initviabranch) ||
         ast_strlen_zero(arg->viasentby) ||
         strcmp(arg->viasentby, sip_pvt_ptr->initviasentby)) {
         /* At this point, this request does not match this Dialog.*/

         /* if methods are different this is just a mismatch */
         if ((sip_pvt_ptr->method != arg->method)) {
            return SIP_REQ_NOT_MATCH;
         }

         /* If RUIs are different, this is a forked request to a separate URI.
          * Returning a mismatch allows this Request to be processed separately. */
         if (sip_uri_cmp(init_ruri, arg->ruri)) {
            /* not a match, request uris are different */
            return SIP_REQ_NOT_MATCH;
         }

         /* Loop/Merge Detected
          *
          * ---Current Matches to Initial Request---
          * request uri
          * Call-id
          * their-tag
          * no totag present
          * method
          * cseq
          *
          * --- Does not Match Initial Request ---
          * Top Via
          *
          * Without the same Via, this can not match our initial transaction for this dialog,
          * but given that this Request matches everything else associated with that initial
          * Request this is most certainly a Forked request in which we have already received
          * part of the fork.
          */
         return SIP_REQ_LOOP_DETECTED;
      }
   } /* end of Request Via check */

   /* Match Authentication Request.
    *
    * A Request with an Authentication header must come back with the
    * same Request URI.  Otherwise it is not a match.
    */
   if ((arg->method != SIP_RESPONSE) &&      /* Must be a Request type to even begin checking this */
      ast_strlen_zero(arg->totag) &&        /* no totag is present to match */
      arg->authentication_present &&        /* Authentication header is present in Request */
      sip_uri_cmp(init_ruri, arg->ruri)) {  /* Compare the Request URI of both the last Request and this new one */

      /* Authentication was provided, but the Request URI did not match the last one on this dialog. */
      return SIP_REQ_NOT_MATCH;
   }

   return SIP_REQ_MATCH;
}

static int method_match	(	enum sipmethod	id,
		const char *	name
	)			[static]

returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send

Definition at line 2902 of file chan_sip.c.

References len(), sip_methods, and text.

Referenced by __sip_autodestruct(), __sip_semi_ack(), and find_sip_method().

{
   int len = strlen(sip_methods[id].text);
   int l_name = name ? strlen(name) : 0;
   /* true if the string is long enough, and ends with whitespace, and matches */
   return (l_name >= len && name[len] < 33 &&
      !strncasecmp(sip_methods[id].text, name, len));
}

static void mwi_event_cb	(	const struct ast_event *	event,
		void *	userdata
	)			[static]

Receive MWI events that we have subscribed to.

Definition at line 13453 of file chan_sip.c.

References ao2_lock, ao2_unlock, and sip_send_mwi_to_peer().

{
   struct sip_peer *peer = userdata;

   ao2_lock(peer);
   sip_send_mwi_to_peer(peer, event, 0);
   ao2_unlock(peer);
}

static void network_change_event_cb	(	const struct ast_event *	event,
		void *	userdata
	)			[static]

Definition at line 13487 of file chan_sip.c.

References ast_debug, ast_sched_add(), and network_change_event_sched_cb().

{
   ast_debug(1, "SIP, got a network change event, renewing all SIP registrations.\n");
   if (network_change_event_sched_id == -1) {
      network_change_event_sched_id = ast_sched_add(sched, 1000, network_change_event_sched_cb, NULL);
   }
}

static int network_change_event_sched_cb

(

const void *

data

)

[static]

Definition at line 13479 of file chan_sip.c.

References sip_send_all_mwi_subscriptions(), and sip_send_all_registers().

{
   network_change_event_sched_id = -1;
   sip_send_all_registers();
   sip_send_all_mwi_subscriptions();
   return 0;
}

static void network_change_event_subscribe

(

void

)

[static]

Definition at line 13462 of file chan_sip.c.

References AST_EVENT_IE_END, AST_EVENT_NETWORK_CHANGE, ast_event_subscribe(), and network_change_event_cb().

{
   if (!network_change_event_subscription) {
      network_change_event_subscription = ast_event_subscribe(AST_EVENT_NETWORK_CHANGE,
         network_change_event_cb,
         "SIP Network Change ",
         NULL, AST_EVENT_IE_END);
   }
}

static void network_change_event_unsubscribe

(

void

)

[static]

Definition at line 13472 of file chan_sip.c.

References ast_event_unsubscribe().

{
   if (network_change_event_subscription) {
      network_change_event_subscription = ast_event_unsubscribe(network_change_event_subscription);
   }
}

static struct sip_proxy* obproxy_get	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)			[static, read]

Get default outbound proxy or global proxy.

Definition at line 2874 of file chan_sip.c.

References append_history, ast_debug, and sip_cfg.

Referenced by __sip_subscribe_mwi_do(), create_addr(), and create_addr_from_peer().

{
   if (peer && peer->outboundproxy) {
      if (sipdebug) {
         ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
      }
      append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
      return peer->outboundproxy;
   }
   if (sip_cfg.outboundproxy.name[0]) {
      if (sipdebug) {
         ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
      }
      append_history(dialog, "OBproxy", "Using global obproxy %s", sip_cfg.outboundproxy.name);
      return &sip_cfg.outboundproxy;
   }
   if (sipdebug) {
      ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
   }
   return NULL;
}

static unsigned int parse_allowed_methods

(

struct sip_request *

req

)

[static]

parse the Allow header to see what methods the endpoint we are communicating with allows.

We parse the allow header on incoming Registrations and save the result to the SIP peer that is registering. When the registration expires, we clear what we know about the peer's allowed methods. When the peer re-registers, we once again parse to see if the list of allowed methods has changed.

For peers that do not register, we parse the first message we receive during a call to see what is allowed, and save the information for the duration of the call.

Parameters:

req

The SIP request we are parsing

Return values:

The

methods allowed

Definition at line 7636 of file chan_sip.c.

References ast_strdupa, ast_strip_quoted(), ast_strlen_zero(), get_header(), and mark_parsed_methods().

Referenced by set_pvt_allowed_methods().

{
   char *allow = ast_strdupa(get_header(req, "Allow"));
   unsigned int allowed_methods = SIP_UNKNOWN;

   if (ast_strlen_zero(allow)) {
      /* I have witnessed that REGISTER requests from Polycom phones do not
       * place the phone's allowed methods in an Allow header. Instead, they place the
       * allowed methods in a methods= parameter in the Contact header.
       */
      char *contact = ast_strdupa(get_header(req, "Contact"));
      char *methods = strstr(contact, ";methods=");

      if (ast_strlen_zero(methods)) {
         /* RFC 3261 states:
          *
          * "The absence of an Allow header field MUST NOT be
          * interpreted to mean that the UA sending the message supports no
          * methods.   Rather, it implies that the UA is not providing any
          * information on what methods it supports."
          *
          * For simplicity, we'll assume that the peer allows all known
          * SIP methods if they have no Allow header. We can then clear out the necessary
          * bits if the peer lets us know that we have sent an unsupported method.
          */
         return UINT_MAX;
      }
      allow = ast_strip_quoted(methods + 9, "\"", "\"");
   }
   mark_parsed_methods(&allowed_methods, allow);
   return allowed_methods;
}

static void parse_copy	(	struct sip_request *	dst,
		const struct sip_request *	src
	)			[static]

Copy SIP request, parse it.

Definition at line 3808 of file chan_sip.c.

References copy_request(), and parse_request().

Referenced by send_request(), and send_response().

{
   copy_request(dst, src);
   parse_request(dst);
}

int parse_minse	(	const char *	p_hdrval,
		int *const	p_interval
	)			[static]

Session-Timers: Function for parsing Min-SE header.

Definition at line 24464 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strlen_zero(), and LOG_WARNING.

Referenced by handle_request_invite(), and proc_422_rsp().

{
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Min-SE header\n");
      return -1;
   }

   *p_interval = 0;
   p_hdrval = ast_skip_blanks(p_hdrval);
   if (!sscanf(p_hdrval, "%30d", p_interval)) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return -1;
   }

   ast_debug(2, "Received Min-SE: %d\n", *p_interval);
   return 0;
}

static void parse_moved_contact	(	struct sip_pvt *	p,
		struct sip_request *	req,
		char **	name,
		char **	number,
		int	set_call_forward
	)			[static]

Parse 302 Moved temporalily response.

Todo:

XXX Doesn't redirect over TLS on sips: uri's. If we get a redirect to a SIPS: uri, this needs to be going back to the dialplan (this is a request for a secure signalling path). Note that transport=tls is deprecated, but we need to support it on incoming requests.

Definition at line 18414 of file chan_sip.c.

References ao2_ref, ast_copy_string(), ast_debug, ast_log(), ast_strdup, ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), find_closing_quote(), get_header(), get_transport(), LOG_NOTICE, pbx_builtin_setvar_helper(), remove_uri_parameters(), set_socket_transport(), and strcasestr().

Referenced by change_redirecting_information().

{
   char contact[SIPBUFSIZE];
   char *contact_name = NULL;
   char *contact_number = NULL;
   char *separator, *trans;
   char *domain;
   enum sip_transport transport = SIP_TRANSPORT_UDP;

   ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
   if ((separator = strchr(contact, ',')))
      *separator = '\0';

   contact_number = get_in_brackets(contact);
   if ((trans = strcasestr(contact_number, ";transport="))) {
      trans += 11;

      if ((separator = strchr(trans, ';')))
         *separator = '\0';

      if (!strncasecmp(trans, "tcp", 3))
         transport = SIP_TRANSPORT_TCP;
      else if (!strncasecmp(trans, "tls", 3))
         transport = SIP_TRANSPORT_TLS;
      else {
         if (strncasecmp(trans, "udp", 3))
            ast_debug(1, "received contact with an invalid transport, '%s'\n", contact_number);
         /* This will assume UDP for all unknown transports */
         transport = SIP_TRANSPORT_UDP;
      }
   }
   contact_number = remove_uri_parameters(contact_number);

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   }

   set_socket_transport(&p->socket, transport);

   if (set_call_forward && ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
      char *host = NULL;
      if (!strncasecmp(contact_number, "sip:", 4))
         contact_number += 4;
      else if (!strncasecmp(contact_number, "sips:", 5))
         contact_number += 5;
      separator = strchr(contact_number, '/');
      if (separator)
         *separator = '\0';
      if ((host = strchr(contact_number, '@'))) {
         *host++ = '\0';
         ast_debug(2, "Found promiscuous redirection to 'SIP/%s::::%s@%s'\n", contact_number, get_transport(transport), host);
         if (p->owner)
            ast_string_field_build(p->owner, call_forward, "SIP/%s::::%s@%s", contact_number, get_transport(transport), host);
      } else {
         ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", get_transport(transport), contact_number);
         if (p->owner)
            ast_string_field_build(p->owner, call_forward, "SIP/::::%s@%s", get_transport(transport), contact_number);
      }
   } else {
      separator = strchr(contact, '@');
      if (separator) {
         *separator++ = '\0';
         domain = separator;
      } else {
         /* No username part */
         domain = contact;
      }
      separator = strchr(contact, '/');   /* WHEN do we hae a forward slash in the URI? */
      if (separator)
         *separator = '\0';

      if (!strncasecmp(contact_number, "sip:", 4))
         contact_number += 4;
      else if (!strncasecmp(contact_number, "sips:", 5))
         contact_number += 5;
      separator = strchr(contact_number, ';');  /* And username ; parameters? */
      if (separator)
         *separator = '\0';
      ast_uri_decode(contact_number);
      if (set_call_forward) {
         ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", contact_number, domain);
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
            ast_string_field_set(p->owner, call_forward, contact_number);
         }
      }
   }

   /* We've gotten the number for the contact, now get the name */

   if (*contact == '\"') {
      contact_name = contact + 1;
      if (!(separator = (char *)find_closing_quote(contact_name, NULL))) {
         ast_log(LOG_NOTICE, "No closing quote on name in Contact header? %s\n", contact);
      }
      *separator = '\0';
   }

   if (name && !ast_strlen_zero(contact_name)) {
      *name = ast_strdup(contact_name);
   }
   if (number) {
      *number = ast_strdup(contact_number);
   }
}

static int parse_ok_contact	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)			[static]

Save contact header for 200 OK on INVITE.

Definition at line 12815 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, get_header(), and TRUE.

Referenced by handle_request_invite(), handle_request_subscribe(), and handle_response_invite().

{
   char contact[SIPBUFSIZE];
   char *c;

   /* Look for brackets */
   ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
   c = get_in_brackets(contact);

   /* Save full contact to call pvt for later bye or re-invite */
   ast_string_field_set(pvt, fullcontact, c);

   /* Save URI for later ACKs, BYE or RE-invites */
   ast_string_field_set(pvt, okcontacturi, c);

   /* We should return false for URI:s we can't handle,
      like tel:, mailto:,ldap: etc */
   return TRUE;      
}

static enum parse_register_result parse_register_contact	(	struct sip_pvt *	pvt,
		struct sip_peer *	p,
		struct sip_request *	req
	)			[static]

Parse contact header and save registration (peer registration).

Todo:

Check NAPTR/SRV if we have not got a port in the URI

Definition at line 12897 of file chan_sip.c.

References ao2_t_link, ao2_t_unlink, ast_apply_ha(), ast_copy_string(), ast_db_put(), ast_debug, ast_log(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_sched_when(), AST_SENSE_ALLOW, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, ast_verbose, copy_socket_data(), destroy_association(), EVENT_FLAG_SYSTEM, expire_register(), FALSE, get_header(), get_transport_str2enum(), LOG_NOTICE, LOG_WARNING, manager_event, ref_peer(), register_peer_exten(), set_socket_transport(), sip_cfg, sip_poke_peer(), strcasestr(), strsep(), TRUE, unref_peer(), VERBOSE_PREFIX_3, and VERBOSITY_ATLEAST.

Referenced by register_verify().

{
   char contact[SIPBUFSIZE];
   char data[SIPBUFSIZE];
   const char *expires = get_header(req, "Expires");
   int expire = atoi(expires);
   char *curi, *domain, *transport;
   int transport_type;
   const char *useragent;
   struct ast_sockaddr oldsin, testsa;

   ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));

   if (ast_strlen_zero(expires)) {  /* No expires header, try look in Contact: */
      char *s = strcasestr(contact, ";expires=");
      if (s) {
         expires = strsep(&s, ";"); /* trim ; and beyond */
         if (sscanf(expires + 9, "%30d", &expire) != 1) {
            expire = default_expiry;
         }
      } else {
         /* Nothing has been specified */
         expire = default_expiry;
      }
   }

   copy_socket_data(&pvt->socket, &req->socket);

   /* Look for brackets */
   curi = contact;
   if (strchr(contact, '<') == NULL)   /* No <, check for ; and strip it */
      strsep(&curi, ";");  /* This is Header options, not URI options */
   curi = get_in_brackets(contact);

   /* if they did not specify Contact: or Expires:, they are querying
      what we currently have stored as their contact address, so return
      it
   */
   if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) {
      /* If we have an active registration, tell them when the registration is going to expire */
      if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact)) {
         pvt->expiry = ast_sched_when(sched, peer->expire);
      }
      return PARSE_REGISTER_QUERY;
   } else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
      /* This means remove all registrations and return OK */
      memset(&peer->addr, 0, sizeof(peer->addr));
      set_socket_transport(&peer->socket, peer->default_outbound_transport);

      AST_SCHED_DEL_UNREF(sched, peer->expire,
            unref_peer(peer, "remove register expire ref"));

      destroy_association(peer);

      register_peer_exten(peer, FALSE);   /* Remove extension from regexten= setting in sip.conf */
      ast_string_field_set(peer, fullcontact, "");
      ast_string_field_set(peer, useragent, "");
      peer->sipoptions = 0;
      peer->lastms = 0;
      peer->portinuri = 0;
      pvt->expiry = 0;

      ast_verb(3, "Unregistered SIP '%s'\n", peer->name);

      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\n", peer->name);
      return PARSE_REGISTER_UPDATE;
   }

   /* Store whatever we got as a contact from the client */
   ast_string_field_set(peer, fullcontact, curi);

   /* For the 200 OK, we should use the received contact */
   ast_string_field_build(pvt, our_contact, "<%s>", curi);

   /* Make sure it's a SIP URL */
   if (parse_uri(curi, "sip:,sips:", &curi, NULL, &domain, &transport)) {
      ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:/sips:) trying to use anyway\n");
   }

   /* handle the transport type specified in Contact header. */
   if (!(transport_type = get_transport_str2enum(transport))) {
      transport_type = pvt->socket.type;
   }

   /* if the peer's socket type is different than the Registration
    * transport type, change it.  If it got this far, it is a
    * supported type, but check just in case */
   if ((peer->socket.type != transport_type) && (peer->transports & transport_type)) {
      set_socket_transport(&peer->socket, transport_type);
   }

   oldsin = peer->addr;

   /* If we were already linked into the peers_by_ip container unlink ourselves so nobody can find us */
   if (!ast_sockaddr_isnull(&peer->addr)) {
      ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
   }

   if (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_RPORT_PRESENT)) {
       /* use the data provided in the Contact header for call routing */
      ast_debug(1, "Store REGISTER's Contact header for call routing.\n");
      /* XXX This could block for a long time XXX */
      /*! \todo Check NAPTR/SRV if we have not got a port in the URI */
      if (ast_sockaddr_resolve_first(&testsa, domain, 0)) {
         ast_log(LOG_WARNING, "Invalid domain '%s'\n", domain);
         ast_string_field_set(peer, fullcontact, "");
         ast_string_field_set(pvt, our_contact, "");
         return PARSE_REGISTER_FAILED;
      }

      /* If we have a port number in the given URI, make sure we do remember to not check for NAPTR/SRV records.
         The domain part is actually a host. */
      peer->portinuri = ast_sockaddr_port(&testsa) ? TRUE : FALSE;

      if (!ast_sockaddr_port(&testsa)) {
         ast_sockaddr_set_port(&testsa,
                    transport_type == SIP_TRANSPORT_TLS ?
                    STANDARD_TLS_PORT : STANDARD_SIP_PORT);
      }

      ast_sockaddr_copy(&peer->addr, &testsa);
   } else {
      /* Don't trust the contact field.  Just use what they came to us
         with */
      ast_debug(1, "Store REGISTER's src-IP:port for call routing.\n");
      peer->addr = pvt->recv;
   }

   /* Check that they're allowed to register at this IP */
   if (ast_apply_ha(sip_cfg.contact_ha, &peer->addr) != AST_SENSE_ALLOW ||
         ast_apply_ha(peer->contactha, &peer->addr) != AST_SENSE_ALLOW) {
      ast_log(LOG_WARNING, "Domain '%s' disallowed by contact ACL (violating IP %s)\n", domain,
         ast_sockaddr_stringify_addr(&testsa));
      ast_string_field_set(peer, fullcontact, "");
      ast_string_field_set(pvt, our_contact, "");
      return PARSE_REGISTER_DENIED;
   }

   /* if the Contact header information copied into peer->addr matches the
    * received address, and the transport types are the same, then copy socket
    * data into the peer struct */
   if ((peer->socket.type == pvt->socket.type) &&
      !ast_sockaddr_cmp(&peer->addr, &pvt->recv)) {
      copy_socket_data(&peer->socket, &pvt->socket);
   }

   /* Now that our address has been updated put ourselves back into the container for lookups */
   ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");

   /* Save SIP options profile */
   peer->sipoptions = pvt->sipoptions;

   if (!ast_strlen_zero(curi) && ast_strlen_zero(peer->username)) {
      ast_string_field_set(peer, username, curi);
   }

   AST_SCHED_DEL_UNREF(sched, peer->expire,
         unref_peer(peer, "remove register expire ref"));

   if (expire > max_expiry) {
      expire = max_expiry;
   }
   if (expire < min_expiry) {
      expire = min_expiry;
   }
   if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      peer->expire = -1;
   } else {
      peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
            ref_peer(peer, "add registration ref"));
      if (peer->expire == -1) {
         unref_peer(peer, "remote registration ref");
      }
   }
   pvt->expiry = expire;
   snprintf(data, sizeof(data), "%s:%d:%s:%s", ast_sockaddr_stringify(&peer->addr),
       expire, peer->username, peer->fullcontact);
   /* Saving TCP connections is useless, we won't be able to reconnect
      XXX WHY???? XXX
      \todo Fix this immediately.
   */
   if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
      ast_db_put("SIP/Registry", peer->name, data);
   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\n", peer->name,  ast_sockaddr_stringify(&peer->addr));

   /* Is this a new IP address for us? */
   if (VERBOSITY_ATLEAST(2) && ast_sockaddr_cmp(&peer->addr, &oldsin)) {
      ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s\n", peer->name,
            ast_sockaddr_stringify(&peer->addr));
   }
   sip_poke_peer(peer, 0);
   register_peer_exten(peer, 1);
   
   /* Save User agent */
   useragent = get_header(req, "User-Agent");
   if (strcasecmp(useragent, peer->useragent)) {
      ast_string_field_set(peer, useragent, useragent);
      ast_verb(4, "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name);
   }
   return PARSE_REGISTER_UPDATE;
}

static int parse_request

(

struct sip_request *

req

)

[static]

Parse a SIP message.

Note:

this function is used both on incoming and outgoing packets

Definition at line 7739 of file chan_sip.c.

References ast_debug, ast_log(), ast_str_strlen(), ast_strlen_zero(), determine_firstline_parts(), and LOG_WARNING.

Referenced by _sip_tcp_helper_thread(), handle_request_do(), initialize_initreq(), and parse_copy().

{
   char *c = req->data->str;
   ptrdiff_t *dst = req->header;
   int i = 0, lim = SIP_MAX_HEADERS - 1;
   unsigned int skipping_headers = 0;
   ptrdiff_t current_header_offset = 0;
   char *previous_header = "";

   req->header[0] = 0;
   req->headers = -1;   /* mark that we are working on the header */
   for (; *c; c++) {
      if (*c == '\r') {    /* remove \r */
         *c = '\0';
      } else if (*c == '\n') {   /* end of this line */
         *c = '\0';
         current_header_offset = (c + 1) - req->data->str;
         previous_header = req->data->str + dst[i];
         if (skipping_headers) {
            /* check to see if this line is blank; if so, turn off
               the skipping flag, so the next line will be processed
               as a body line */
            if (ast_strlen_zero(previous_header)) {
               skipping_headers = 0;
            }
            dst[i] = current_header_offset; /* record start of next line */
            continue;
         }
         if (sipdebug) {
            ast_debug(4, "%7s %2d [%3d]: %s\n",
                 req->headers < 0 ? "Header" : "Body",
                 i, (int) strlen(previous_header), previous_header);
         }
         if (ast_strlen_zero(previous_header) && req->headers < 0) {
            req->headers = i; /* record number of header lines */
            dst = req->line;  /* start working on the body */
            i = 0;
            lim = SIP_MAX_LINES - 1;
         } else { /* move to next line, check for overflows */
            if (i++ == lim) {
               /* if we're processing headers, then skip any remaining
                  headers and move on to processing the body, otherwise
                  we're done */
               if (req->headers != -1) {
                  break;
               } else {
                  req->headers = i;
                  dst = req->line;
                  i = 0;
                  lim = SIP_MAX_LINES - 1;
                  skipping_headers = 1;
               }
            }
         }
         dst[i] = current_header_offset; /* record start of next line */
      }
   }

   /* Check for last header or body line without CRLF. The RFC for SDP requires CRLF,
      but since some devices send without, we'll be generous in what we accept. However,
      if we've already reached the maximum number of lines for portion of the message
      we were parsing, we can't accept any more, so just ignore it.
   */
   previous_header = req->data->str + dst[i];
   if ((i < lim) && !ast_strlen_zero(previous_header)) {
      if (sipdebug) {
         ast_debug(4, "%7s %2d [%3d]: %s\n",
              req->headers < 0 ? "Header" : "Body",
              i, (int) strlen(previous_header), previous_header );
      }
      i++;
   }

   /* update count of header or body lines */
   if (req->headers >= 0) {   /* we are in the body */
      req->lines = i;
   } else {       /* no body */
      req->headers = i;
      req->lines = 0;
      /* req->data->used will be a NULL byte */
      req->line[0] = ast_str_strlen(req->data);
   }

   if (*c) {
      ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
   }

   /* Split up the first line parts */
   return determine_firstline_parts(req);
}

int parse_session_expires	(	const char *	p_hdrval,
		int *const	p_interval,
		enum st_refresher *const	p_ref
	)			[static]

Session-Timers: Function for parsing Session-Expires header.

Definition at line 24484 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strdupa, ast_strlen_zero(), LOG_WARNING, and strsep().

Referenced by handle_request_invite(), and handle_response_invite().

{
   char *p_token;
   int  ref_idx;
   char *p_se_hdr;

   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Session-Expires header\n");
      return -1;
   }

   *p_ref = SESSION_TIMER_REFRESHER_AUTO;
   *p_interval = 0;

   p_se_hdr = ast_strdupa(p_hdrval);
   p_se_hdr = ast_skip_blanks(p_se_hdr);

   while ((p_token = strsep(&p_se_hdr, ";"))) {
      p_token = ast_skip_blanks(p_token);
      if (!sscanf(p_token, "%30d", p_interval)) {
         ast_log(LOG_WARNING, "Parsing of Session-Expires failed\n");
         return -1;
      }

      ast_debug(2, "Session-Expires: %d\n", *p_interval);

      if (!p_se_hdr)
         continue;

      p_se_hdr = ast_skip_blanks(p_se_hdr);
      ref_idx = strlen("refresher=");
      if (!strncasecmp(p_se_hdr, "refresher=", ref_idx)) {
         p_se_hdr += ref_idx;
         p_se_hdr = ast_skip_blanks(p_se_hdr);

         if (!strncasecmp(p_se_hdr, "uac", strlen("uac"))) {
            *p_ref = SESSION_TIMER_REFRESHER_UAC;
            ast_debug(2, "Refresher: UAC\n");
         } else if (!strncasecmp(p_se_hdr, "uas", strlen("uas"))) {
            *p_ref = SESSION_TIMER_REFRESHER_UAS;
            ast_debug(2, "Refresher: UAS\n");
         } else {
            ast_log(LOG_WARNING, "Invalid refresher value %s\n", p_se_hdr);
            return -1;
         }
         break;
      }
   }
   return 0;
}

static int peer_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Note:

The only member of the peer used here is the name field

Definition at line 27974 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

{
   struct sip_peer *peer = obj, *peer2 = arg;

   return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
}

static int peer_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 15688 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and ast_cli_args::fd.

Referenced by sip_show_objects().

{
   struct sip_peer *peer = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   struct ast_cli_args *a = (struct ast_cli_args *) arg;
   
   ast_cli(a->fd, "name: %s\ntype: peer\nobjflags: %d\nrefcount: %d\n\n",
      peer->name, 0, refc);
   return 0;
}

static int peer_hash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Note:

The only member of the peer used here is the name field

Definition at line 27964 of file chan_sip.c.

References ast_str_case_hash().

{
   const struct sip_peer *peer = obj;

   return ast_str_case_hash(peer->name);
}

static int peer_ipcmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)			[static]

Match Peers by IP and Port number.

This function has two modes.

• If the peer arg does not have INSECURE_PORT set, then we will only return a match for a peer that matches both the IP and port.
• If the peer arg does have the INSECURE_PORT flag set, then we will only return a match for a peer that matches the IP and has insecure=port in its configuration.

This callback will be used twice when doing peer matching. There is a first pass for full IP+port matching, and a second pass in case there is a match that meets the insecure=port criteria.

Note:

Connections coming in over TCP or TLS should never be matched by port.

the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.

Definition at line 28022 of file chan_sip.c.

References ast_sockaddr_cmp_addr(), ast_sockaddr_port, ast_test_flag, CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_peer *peer = obj, *peer2 = arg;

   if (ast_sockaddr_cmp_addr(&peer->addr, &peer2->addr)) {
      /* IP doesn't match */
      return 0;
   }

   /* We matched the IP, check to see if we need to match by port as well. */
   if ((peer->transports & peer2->transports) & (SIP_TRANSPORT_TLS | SIP_TRANSPORT_TCP)) {
      /* peer matching on port is not possible with TCP/TLS */
      return CMP_MATCH | CMP_STOP;
   } else if (ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
      /* We are allowing match without port for peers configured that
       * way in this pass through the peers. */
      return ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) ?
            (CMP_MATCH | CMP_STOP) : 0;
   }

   /* Now only return a match if the port matches, as well. */
   return ast_sockaddr_port(&peer->addr) == ast_sockaddr_port(&peer2->addr) ?
         (CMP_MATCH | CMP_STOP) : 0;
}

static int peer_iphash_cb	(	const void *	obj,
		const int	flags
	)			[static]

Hash function based on the the peer's ip address. For IPv6, we use the end of the address.

Todo:

Find a better hashing function

Definition at line 27986 of file chan_sip.c.

References ast_log(), ast_sockaddr_hash(), ast_sockaddr_isnull(), and LOG_ERROR.

Referenced by load_module().

{
   const struct sip_peer *peer = obj;
   int ret = 0;

   if (ast_sockaddr_isnull(&peer->addr)) {
      ast_log(LOG_ERROR, "Empty address\n");
   }

   ret = ast_sockaddr_hash(&peer->addr);

   if (ret < 0) {
      ret = -ret;
   }

   return ret;
}

static int peer_is_marked	(	void *	peerobj,
		void *	arg,
		int	flags
	)			[static]

Definition at line 2627 of file chan_sip.c.

References CMP_MATCH.

Referenced by unlink_marked_peers_from_tables().

{
   struct sip_peer *peer = peerobj;
   return peer->the_mark ? CMP_MATCH : 0;
}

static void peer_mailboxes_to_str	(	struct ast_str **	mailbox_str,
		struct sip_peer *	peer
	)			[static]

list peer mailboxes to CLI

Definition at line 16171 of file chan_sip.c.

References AST_LIST_NEXT, AST_LIST_TRAVERSE, ast_str_append(), ast_strlen_zero(), mailbox, and S_OR.

Referenced by _sip_show_peer(), function_sippeer(), show_channels_cb(), and sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      ast_str_append(mailbox_str, 0, "%s%s%s%s",
         mailbox->mailbox,
         ast_strlen_zero(mailbox->context) ? "" : "@",
         S_OR(mailbox->context, ""),
         AST_LIST_NEXT(mailbox, entry) ? "," : "");
   }
}

static int peer_markall_func	(	void *	device,
		void *	arg,
		int	flags
	)			[static]

Definition at line 26240 of file chan_sip.c.

Referenced by reload_config().

{
   struct sip_peer *peer = device;
   peer->the_mark = 1;
   return 0;
}

static int peer_status	(	struct sip_peer *	peer,
		char *	status,
		int	statuslen
	)			[static]

Definition at line 15278 of file chan_sip.c.

References ast_copy_string().

{
   int res = 0;
   if (peer->maxms) {
      if (peer->lastms < 0) {
         ast_copy_string(status, "UNREACHABLE", statuslen);
      } else if (peer->lastms > peer->maxms) {
         snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms);
         res = 1;
      } else if (peer->lastms) {
         snprintf(status, statuslen, "OK (%d ms)", peer->lastms);
         res = 1;
      } else {
         ast_copy_string(status, "UNKNOWN", statuslen);
      }
   } else {
      ast_copy_string(status, "Unmonitored", statuslen);
      /* Checking if port is 0 */
      res = -1;
   }
   return res;
}

int peercomparefunc	(	const void *	a,
		const void *	b
	)

Definition at line 15506 of file chan_sip.c.

Referenced by _sip_show_peers().

{
   struct sip_peer **ap = (struct sip_peer **)a;
   struct sip_peer **bp = (struct sip_peer **)b;
   return strcmp((*ap)->name, (*bp)->name);
}

static int peers_data_provider_get	(	const struct ast_data_search *	search,
		struct ast_data *	data_root
	)			[static]

Definition at line 28252 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_lock, ao2_ref, ao2_unlock, ARRAY_LEN, ast_cdr_flags2str(), ast_data_add_bool(), ast_data_add_codecs(), ast_data_add_int(), ast_data_add_node(), ast_data_add_str(), ast_data_add_structure, ast_data_remove_node(), ast_data_search_match(), ast_describe_caller_presentation(), AST_LIST_TRAVERSE, get_transport_list(), mailbox, text, and transfermode2str().

{
   struct sip_peer *peer;
   struct ao2_iterator i;
   struct ast_data *data_peer, *data_peer_mailboxes = NULL, *data_peer_mailbox, *enum_node;
   struct ast_data *data_sip_options;
   int total_mailboxes, x;
   struct sip_mailbox *mailbox;

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_iterator_next(&i))) {
      ao2_lock(peer);

      data_peer = ast_data_add_node(data_root, "peer");
      if (!data_peer) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }

      ast_data_add_structure(sip_peer, data_peer, peer);

      /* transfer mode */
      enum_node = ast_data_add_node(data_peer, "allowtransfer");
      if (!enum_node) {
         continue;
      }
      ast_data_add_str(enum_node, "text", transfermode2str(peer->allowtransfer));
      ast_data_add_int(enum_node, "value", peer->allowtransfer);

      /* transports */
      ast_data_add_str(data_peer, "transports", get_transport_list(peer->transports));

      /* peer type */
      if ((peer->type & SIP_TYPE_USER) && (peer->type & SIP_TYPE_PEER)) {
         ast_data_add_str(data_peer, "type", "friend");
      } else if (peer->type & SIP_TYPE_PEER) {
         ast_data_add_str(data_peer, "type", "peer");
      } else if (peer->type & SIP_TYPE_USER) {
         ast_data_add_str(data_peer, "type", "user");
      }

      /* mailboxes */
      total_mailboxes = 0;
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         if (!total_mailboxes) {
            data_peer_mailboxes = ast_data_add_node(data_peer, "mailboxes");
            if (!data_peer_mailboxes) {
               break;
            }
            total_mailboxes++;
         }

         data_peer_mailbox = ast_data_add_node(data_peer_mailboxes, "mailbox");
         if (!data_peer_mailbox) {
            continue;
         }
         ast_data_add_str(data_peer_mailbox, "mailbox", mailbox->mailbox);
         ast_data_add_str(data_peer_mailbox, "context", mailbox->context);
      }

      /* amaflags */
      enum_node = ast_data_add_node(data_peer, "amaflags");
      if (!enum_node) {
         continue;
      }
      ast_data_add_int(enum_node, "value", peer->amaflags);
      ast_data_add_str(enum_node, "text", ast_cdr_flags2str(peer->amaflags));

      /* sip options */
      data_sip_options = ast_data_add_node(data_peer, "sipoptions");
      if (!data_sip_options) {
         continue;
      }
      for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
         ast_data_add_bool(data_sip_options, sip_options[x].text, peer->sipoptions & sip_options[x].id);
      }

      /* callingpres */
      enum_node = ast_data_add_node(data_peer, "callingpres");
      if (!enum_node) {
         continue;
      }
      ast_data_add_int(enum_node, "value", peer->callingpres);
      ast_data_add_str(enum_node, "text", ast_describe_caller_presentation(peer->callingpres));

      /* codecs */
      ast_data_add_codecs(data_peer, "codecs", peer->capability);

      if (!ast_data_search_match(search, data_peer)) {
         ast_data_remove_node(data_root, data_peer);
      }

      ao2_unlock(peer);
      ao2_ref(peer, -1);
   }
   ao2_iterator_destroy(&i);

   return 0;
}

unsigned int port_str2int	(	const char *	pt,
		unsigned int	standard
	)

converts ascii port to int representation. If no pt buffer is provided or the pt has errors when being converted to an int value, the port provided as the standard is used.

Definition at line 2863 of file chan_sip.c.

References ast_strlen_zero().

Referenced by build_peer(), and reload_config().

{
   int port = standard;
   if (ast_strlen_zero(pt) || (sscanf(pt, "%30d", &port) != 1) || (port < 1) || (port > 65535)) {
      port = standard;
   }

   return port;
}

static void print_codec_to_cli	(	int	fd,
		struct ast_codec_pref *	pref
	)			[static]

Print codec list from preference to CLI/manager.

Definition at line 16008 of file chan_sip.c.

References ast_cli(), ast_codec_pref_index(), ast_getformatname(), and ast_codec_pref::framing.

Referenced by _sip_show_peer(), _skinny_show_line(), sip_show_settings(), and sip_show_user().

{
   int x;
   format_t codec;

   for(x = 0; x < 64 ; x++) {
      codec = ast_codec_pref_index(pref, x);
      if (!codec)
         break;
      ast_cli(fd, "%s", ast_getformatname(codec));
      ast_cli(fd, ":%d", pref->framing[x]);
      if (x < 31 && ast_codec_pref_index(pref, x + 1))
         ast_cli(fd, ",");
   }
   if (!x)
      ast_cli(fd, "none");
}

static void print_group	(	int	fd,
		ast_group_t	group,
		int	crlf
	)			[static]

Print call group and pickup group.

Definition at line 15738 of file chan_sip.c.

References ast_cli(), and ast_print_group().

Referenced by _sip_show_peer(), and sip_show_user().

{
   char buf[256];
   ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) );
}

static void proc_422_rsp	(	struct sip_pvt *	p,
		struct sip_request *	rsp
	)			[static]

Handle 422 response to INVITE with session-timer requested.

Session-Timers: An INVITE originated by Asterisk that asks for session-timers support from the UAS can result into a 422 response. This is how a UAS or an intermediary proxy server tells Asterisk that the session refresh interval offered by Asterisk is too low for them. The proc_422_rsp() function handles a 422 response. It extracts the Min-SE header that comes back in 422 and sends a new INVITE accordingly.

Definition at line 24543 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), get_header(), LOG_WARNING, parse_minse(), and transmit_invite().

Referenced by handle_response_invite().

{
   int rtn;
   const char *p_hdrval;
   int minse;

   p_hdrval = get_header(rsp, "Min-SE");
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "422 response without a Min-SE header %s\n", p_hdrval);
      return;
   }
   rtn = parse_minse(p_hdrval, &minse);
   if (rtn != 0) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return;
   }
   p->stimer->st_interval = minse;
   transmit_invite(p, SIP_INVITE, 1, 2, NULL);
}

static int proc_session_timer

(

const void *

vp

)

[static]

Session-Timers: Process session refresh timeout event.

Definition at line 24383 of file chan_sip.c.

References ast_channel_trylock, ast_channel_unlock, ast_debug, ast_log(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, FALSE, LOG_ERROR, LOG_WARNING, sip_pvt_lock, sip_pvt_unlock, stop_session_timer(), transmit_reinvite_with_sdp(), and TRUE.

Referenced by start_session_timer().

{
   struct sip_pvt *p = (struct sip_pvt *) vp;
   int sendreinv = FALSE;
   int res = 0;

   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in proc_session_timer - %s\n", p->callid);
      goto return_unref;
   }

   ast_debug(2, "Session timer expired: %d - %s\n", p->stimer->st_schedid, p->callid);

   if (!p->owner) {
      goto return_unref;
   }

   if ((p->stimer->st_active != TRUE) || (p->owner->_state != AST_STATE_UP)) {
      goto return_unref;
   }

   switch (p->stimer->st_ref) {
   case SESSION_TIMER_REFRESHER_UAC:
      if (p->outgoing_call == TRUE) {
         sendreinv = TRUE;
      }
      break;
   case SESSION_TIMER_REFRESHER_UAS:
      if (p->outgoing_call != TRUE) {
         sendreinv = TRUE;
      }
      break;
   default:
      ast_log(LOG_ERROR, "Unknown session refresher %d\n", p->stimer->st_ref);
      goto return_unref;
   }

   if (sendreinv == TRUE) {
      res = 1;
      transmit_reinvite_with_sdp(p, FALSE, TRUE);
   } else {
      p->stimer->st_expirys++;
      if (p->stimer->st_expirys >= 2) {
         if (p->stimer->quit_flag) {
            goto return_unref;
         }
         ast_log(LOG_WARNING, "Session-Timer expired - %s\n", p->callid);
         sip_pvt_lock(p);
         while (p->owner && ast_channel_trylock(p->owner)) {
            sip_pvt_unlock(p);
            usleep(1);
            if (p->stimer && p->stimer->quit_flag) {
               goto return_unref;
            }
            sip_pvt_lock(p);
         }

         ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
         ast_channel_unlock(p->owner);
         sip_pvt_unlock(p);
      }
   }

return_unref:
   if (!res) {
      /* An error occurred.  Stop session timer processing */
      if (p->stimer) {
         p->stimer->st_schedid = -1;
         stop_session_timer(p);
      }

      /* If we are not asking to be rescheduled, then we need to release our
       * reference to the dialog. */
      dialog_unref(p, "removing session timer ref");
   }

   return res;
}

static int process_crypto	(	struct sip_pvt *	p,
		struct ast_rtp_instance *	rtp,
		struct sip_srtp **	srtp,
		const char *	a
	)			[static]

Definition at line 27825 of file chan_sip.c.

References ast_debug, ast_log(), ast_set_flag, ast_test_flag, FALSE, LOG_WARNING, setup_srtp(), and TRUE.

Referenced by process_sdp().

{
   if (strncasecmp(a, "crypto:", 7)) {
      return FALSE;
   }
   if (!*srtp) {
      if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
         return FALSE;
      }

      if (setup_srtp(srtp) < 0) {
         return FALSE;
      }
   }

   /* For now, when we receive an INVITE just take the first successful crypto line */
   if ((*srtp)->crypto && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      ast_debug(3, "We've already processed a crypto attribute, skipping '%s'\n", a);
      return FALSE;
   }

   if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
      return FALSE;
   }

   if (sdp_crypto_process((*srtp)->crypto, a, rtp) < 0) {
      return FALSE;
   }

   ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);

   return TRUE;
}

static void process_request_queue	(	struct sip_pvt *	p,
		int *	recount,
		int *	nounlock
	)			[static]

Definition at line 23643 of file chan_sip.c.

References ast_debug, ast_free, AST_LIST_REMOVE_HEAD, and handle_incoming().

Referenced by handle_request_do(), and scheduler_process_request_queue().

{
   struct sip_request *req;

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      if (handle_incoming(p, req, &p->recv, recount, nounlock) == -1) {
         /* Request failed */
         ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
      }
      ast_free(req);
   }
}

static int process_sdp	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	t38action
	)			[static]

Process SIP SDP offer, select formats and activate RTP channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().

< RTP Audio host IP

< RTP video host IP

< RTP text host IP

< UDPTL host ip

< RTP Audio port number

< RTP Video port number

< RTP Text port number

< UDPTL Image port number

Definition at line 7993 of file chan_sip.c.

References append_history, ast_async_goto(), ast_channel_lock, ast_channel_unlock, ast_clear_flag, ast_codec_choose(), AST_CONTROL_HOLD, AST_CONTROL_UNHOLD, ast_copy_string(), ast_debug, ast_exists_extension(), AST_FORMAT_AUDIO_MASK, AST_FORMAT_T140RED, ast_getformatname_multiple(), ast_log(), ast_manager_event, ast_null_frame, ast_queue_control(), ast_queue_control_data(), ast_queue_frame(), ast_rtp_codecs_payload_formats(), ast_rtp_codecs_payloads_clear(), ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_set_m_type(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_instance_get_remote_address(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_remote_address(), ast_rtp_instance_stop(), ast_rtp_lookup_mime_multiple2(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_rtp_red_init(), ast_set_flag, ast_set_read_format(), ast_set_write_format(), ast_skip_blanks(), ast_sockaddr_isnull(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_str_alloca, ast_strlen_zero(), ast_test_flag, ast_udptl_get_far_max_datagram(), ast_udptl_set_error_correction_scheme(), ast_udptl_set_far_max_datagram(), ast_udptl_set_peer(), ast_udptl_stop(), ast_verbose, change_t38_state(), debug, EVENT_FLAG_CALL, FALSE, get_sdp_iterate(), get_sdp_line(), len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, pbx_builtin_setvar_helper(), process_crypto(), process_sdp_a_audio(), process_sdp_a_image(), process_sdp_a_sendonly(), process_sdp_a_text(), process_sdp_a_video(), process_sdp_c(), process_sdp_o(), S_COR, S_OR, sip_cfg, sip_debug_test_pvt(), sip_peer_hold(), text, TRUE, type, UDPTL_ERROR_CORRECTION_NONE, value, and VERBOSE_PREFIX_2.

{
   /* Iterators for SDP parsing */
   int start = req->sdp_start;
   int next = start;
   int iterator = start;

   /* Temporary vars for SDP parsing */
   char type = '\0';
   const char *value = NULL;
   const char *m = NULL;           /* SDP media offer */
   const char *nextm = NULL;
   int len = -1;

   /* Host information */
   struct ast_sockaddr sessionsa;
   struct ast_sockaddr audiosa;
   struct ast_sockaddr videosa;
   struct ast_sockaddr textsa;
   struct ast_sockaddr imagesa;
   struct ast_sockaddr *sa = NULL;  /*!< RTP Audio host IP */
   struct ast_sockaddr *vsa = NULL; /*!< RTP video host IP */
   struct ast_sockaddr *tsa = NULL; /*!< RTP text host IP */
   struct ast_sockaddr *isa = NULL;     /*!< UDPTL host ip */
   int portno = -1;     /*!< RTP Audio port number */
   int vportno = -1;    /*!< RTP Video port number */
   int tportno = -1;    /*!< RTP Text port number */
   int udptlportno = -1;      /*!< UDPTL Image port number */

   /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */ 
   format_t peercapability = 0, vpeercapability = 0, tpeercapability = 0;
   int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;

   struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
   format_t newjointcapability;           /* Negotiated capability */
   format_t newpeercapability;
   int newnoncodeccapability;

   const char *codecs;
   int codec;

   /* SRTP */
   int secure_audio = FALSE;
   int secure_video = FALSE;

   /* Others */
   int sendonly = -1;
   int vsendonly = -1;
   int numberofports;
   int numberofmediastreams = 0;
   int last_rtpmap_codec = 0;
   int red_data_pt[10];    /* For T.140 red */
   int red_num_gen = 0;    /* For T.140 red */
   char red_fmtp[100] = "empty"; /* For T.140 red */
   int debug = sip_debug_test_pvt(p);

   /* START UNKNOWN */
   char buf[SIPBUFSIZE];
   /* END UNKNOWN */

   /* Initial check */
   if (!p->rtp) {
      ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
      return -1;
   }

   /* Make sure that the codec structures are all cleared out */
   ast_rtp_codecs_payloads_clear(&newaudiortp, NULL);
   ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
   ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);

   /* Update our last rtprx when we receive an SDP, too */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   memset(p->offered_media, 0, sizeof(p->offered_media));


   /* default: novideo and notext set */
   p->novideo = TRUE;
   p->notext = TRUE;

   if (p->vrtp) {
      ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
   }

   if (p->trtp) {
      ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
   }

   /* Scan for the first media stream (m=) line to limit scanning of globals */
   nextm = get_sdp_iterate(&next, req, "m");
   if (ast_strlen_zero(nextm)) {
      ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n");
      return -1;
   }

   /* Scan session level SDP parameters (lines before first media stream) */
   while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
      int processed = FALSE;
      switch (type) {
      case 'o':
         /* If we end up receiving SDP that doesn't actually modify the session we don't want to treat this as a fatal
          * error. We just want to ignore the SDP and let the rest of the packet be handled as normal.
          */
         if (!process_sdp_o(value, p))
            return (p->session_modify == FALSE) ? 0 : -1;
         break;
      case 'c':
         if (process_sdp_c(value, &sessionsa)) {
            processed = TRUE;
            sa = &sessionsa;
            vsa = sa;
            tsa = sa;
            isa = sa;
         }
         break;
      case 'a':
         if (process_sdp_a_sendonly(value, &sendonly)) {
            processed = TRUE;
            vsendonly = sendonly;
         }
         else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_image(value, p))
            processed = TRUE;
         break;
      }

      ast_debug(3, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED.");
   }



   /* Scan media stream (m=) specific parameters loop */
   while (!ast_strlen_zero(nextm)) {
      int audio = FALSE;
      int video = FALSE;
      int image = FALSE;
      int text = FALSE;
      char protocol[5] = {0,};
      int x;

      numberofports = 1;
      len = -1;
      start = next;
      m = nextm;
      iterator = next;
      nextm = get_sdp_iterate(&next, req, "m");

      /* Search for audio media definition */
      if ((sscanf(m, "audio %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
          (sscanf(m, "audio %30u RTP/%4s %n", &x, protocol, &len) == 2 && len > 0)) {
         if (!strcmp(protocol, "SAVP")) {
            secure_audio = 1;
         } else if (strcmp(protocol, "AVP")) {
            ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
            continue;
         }
         audio = TRUE;
         p->offered_media[SDP_AUDIO].offered = TRUE;
         numberofmediastreams++;
         portno = x;

         /* Scan through the RTP payload types specified in a "m=" line: */
         codecs = m + len;
         ast_copy_string(p->offered_media[SDP_AUDIO].codecs, codecs, sizeof(p->offered_media[SDP_AUDIO].codecs));
         for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP audio format %d\n", codec);
            
            ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
         }
      /* Search for video media definition */
      } else if ((sscanf(m, "video %30u/%30u RTP/%4s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
            (sscanf(m, "video %30u RTP/%4s %n", &x, protocol, &len) == 2 && len >= 0)) {
         if (!strcmp(protocol, "SAVP")) {
            secure_video = 1;
         } else if (strcmp(protocol, "AVP")) {
            ast_log(LOG_WARNING, "unknown SDP media protocol in offer: %s\n", protocol);
            continue;
         }
         video = TRUE;
         p->novideo = FALSE;
         p->offered_media[SDP_VIDEO].offered = TRUE;
         numberofmediastreams++;
         vportno = x;

         /* Scan through the RTP payload types specified in a "m=" line: */
         codecs = m + len;
         ast_copy_string(p->offered_media[SDP_VIDEO].codecs, codecs, sizeof(p->offered_media[SDP_VIDEO].codecs));
         for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP video format %d\n", codec);
            ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
         }
      /* Search for text media definition */
      } else if ((sscanf(m, "text %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
            (sscanf(m, "text %30u RTP/AVP %n", &x, &len) == 1 && len > 0)) {
         text = TRUE;
         p->notext = FALSE;
         p->offered_media[SDP_TEXT].offered = TRUE;
         numberofmediastreams++;
         tportno = x;

         /* Scan through the RTP payload types specified in a "m=" line: */
         codecs = m + len;
         ast_copy_string(p->offered_media[SDP_TEXT].codecs, codecs, sizeof(p->offered_media[SDP_TEXT].codecs));
         for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
            if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
               ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
               return -1;
            }
            if (debug)
               ast_verbose("Found RTP text format %d\n", codec);
            ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
         }
      /* Search for image media definition */
      } else if (p->udptl && ((sscanf(m, "image %30u udptl t38%n", &x, &len) == 1 && len > 0) ||
               (sscanf(m, "image %30u UDPTL t38%n", &x, &len) == 1 && len > 0) )) {
         image = TRUE;
         if (debug)
            ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
         p->offered_media[SDP_IMAGE].offered = TRUE;
         udptlportno = x;
         numberofmediastreams++;

         if (p->t38.state != T38_ENABLED) {
            memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));

            /* default EC to none, the remote end should
             * respond with the EC they want to use */
            ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
         }
      } else {
         ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
         continue;
      }

      /* Check for number of ports */
      if (numberofports > 1)
         ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
      
      /* Media stream specific parameters */
      while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
         int processed = FALSE;

         switch (type) {
         case 'c':
            if (audio) {
               if (process_sdp_c(value, &audiosa)) {
                  processed = TRUE;
                  sa = &audiosa;
               }
            } else if (video) {
               if (process_sdp_c(value, &videosa)) {
                  processed = TRUE;
                  vsa = &videosa;
               }
            } else if (text) {
               if (process_sdp_c(value, &textsa)) {
                  processed = TRUE;
                  tsa = &textsa;
               }
            } else if (image) {
               if (process_sdp_c(value, &imagesa)) {
                  processed = TRUE;
                  isa = &imagesa;
               }
            }
            break;
         case 'a':
            /* Audio specific scanning */
            if (audio) {
               if (process_sdp_a_sendonly(value, &sendonly))
                  processed = TRUE;
               else if (process_crypto(p, p->rtp, &p->srtp, value))
                  processed = TRUE;
               else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
                  processed = TRUE;
            }
            /* Video specific scanning */
            else if (video) {
               if (process_sdp_a_sendonly(value, &vsendonly))
                  processed = TRUE;
               else if (process_crypto(p, p->vrtp, &p->vsrtp, value))
                  processed = TRUE;
               else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
                  processed = TRUE;
            }
            /* Text (T.140) specific scanning */
            else if (text) {
               if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
                  processed = TRUE;
               else if (process_crypto(p, p->trtp, &p->tsrtp, value))
                  processed = TRUE;
            }
            /* Image (T.38 FAX) specific scanning */
            else if (image) {
               if (process_sdp_a_image(value, p))
                  processed = TRUE;
            }
            break;
         }

         ast_debug(3, "Processing media-level (%s) SDP %c=%s... %s\n",
               (audio == TRUE)? "audio" : (video == TRUE)? "video" : "image",
               type, value,
               (processed == TRUE)? "OK." : "UNSUPPORTED.");
      }
   }


   /* Sanity checks */
   if (!sa && !vsa && !tsa && !isa) {
      ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n");
      return -1;
   }

   if (portno == -1 && vportno == -1 && udptlportno == -1  && tportno == -1) {
      /* No acceptable offer found in SDP  - we have no ports */
      /* Do not change RTP or VRTP if this is a re-invite */
      ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
      return -2;
   }

   if (numberofmediastreams > 3) {
      /* We have too many fax, audio and/or video and/or text media streams, fail this offer */
      ast_log(LOG_WARNING, "Faling due to too many media streams\n");
      return -3;
   }

   if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
      ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
      return -4;
   }

   if (!secure_audio && p->srtp) {
      ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
      return -4;
   }

   if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
      ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
      return -4;
   }

   if (!p->novideo && !secure_video && p->vsrtp) {
      ast_log(LOG_WARNING, "We are requesting SRTP, but they responded without it!\n");
      return -4;
   }

   if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
      ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
      return -4;
   }

   if (udptlportno == -1) {
      change_t38_state(p, T38_DISABLED);
   }

   /* Now gather all of the codecs that we are asked for: */
   ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
   ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
   ast_rtp_codecs_payload_formats(&newtextrtp, &tpeercapability, &tpeernoncodeccapability);

   newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
   newpeercapability = (peercapability | vpeercapability | tpeercapability);
   newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;

   if (debug) {
      /* shame on whoever coded this.... */
      char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];

      ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
             ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
             ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
             ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
             ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
             ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
   }
   if (debug) {
      struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
      struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
      struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);

      ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
             ast_rtp_lookup_mime_multiple2(s1, p->noncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0));
   }
   if (!newjointcapability && (portno != -1)) {
      ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
      /* Do NOT Change current setting */
      return -1;
   }

   /* Setup audio address and port */
   if (p->rtp) {
      if (portno > 0) {
         ast_sockaddr_set_port(sa, portno);
         ast_rtp_instance_set_remote_address(p->rtp, sa);
         if (debug) {
            ast_verbose("Peer audio RTP is at port %s\n",
                   ast_sockaddr_stringify(sa));
         }
         /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
            they are acceptable */
         p->jointcapability = newjointcapability;                /* Our joint codec profile for this call */
         p->peercapability = newpeercapability;                  /* The other sides capability in latest offer */
         p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */

         if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
            p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
         }

         ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);

         if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
            ast_clear_flag(&p->flags[0], SIP_DTMF);
            if (newnoncodeccapability & AST_RTP_DTMF) {
               /* XXX Would it be reasonable to drop the DSP at this point? XXX */
               ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
               /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
               ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
               ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
            } else {
               ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
            }
         }
      } else if (udptlportno > 0) {
         if (debug)
            ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
      } else {
         ast_rtp_instance_stop(p->rtp);
         if (debug)
            ast_verbose("Peer doesn't provide audio\n");
      }
   }

   /* Setup video address and port */
   if (p->vrtp) {
      if (vportno > 0) {
         ast_sockaddr_set_port(vsa, vportno);
         ast_rtp_instance_set_remote_address(p->vrtp, vsa);
         if (debug) {
            ast_verbose("Peer video RTP is at port %s\n",
                   ast_sockaddr_stringify(vsa));
         }
         ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
      } else {
         ast_rtp_instance_stop(p->vrtp);
         if (debug)
            ast_verbose("Peer doesn't provide video\n");
      }
   }

   /* Setup text address and port */
   if (p->trtp) {
      if (tportno > 0) {
         ast_sockaddr_set_port(tsa, tportno);
         ast_rtp_instance_set_remote_address(p->trtp, tsa);
         if (debug) {
            ast_verbose("Peer T.140 RTP is at port %s\n",
                   ast_sockaddr_stringify(tsa));
         }
         if ((p->jointcapability & AST_FORMAT_T140RED)) {
            p->red = 1;
            ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
         } else {
            p->red = 0;
         }
         ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
      } else {
         ast_rtp_instance_stop(p->trtp);
         if (debug)
            ast_verbose("Peer doesn't provide T.140\n");
      }
   }
   /* Setup image address and port */
   if (p->udptl) {
      if (udptlportno > 0) {
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
            ast_rtp_instance_get_remote_address(p->rtp, isa);
            if (!ast_sockaddr_isnull(isa) && debug) {
               ast_debug(1, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_sockaddr_stringify(isa));
            }
         }
         ast_sockaddr_set_port(isa, udptlportno);
         ast_udptl_set_peer(p->udptl, isa);
         if (debug)
            ast_debug(1,"Peer T.38 UDPTL is at port %s\n", ast_sockaddr_stringify(isa));

         /* verify the far max ifp can be calculated. this requires far max datagram to be set. */
         if (!ast_udptl_get_far_max_datagram(p->udptl)) {
            /* setting to zero will force a default if none was provided by the SDP */
            ast_udptl_set_far_max_datagram(p->udptl, 0);
         }

         /* Remote party offers T38, we need to update state */
         if ((t38action == SDP_T38_ACCEPT) &&
             (p->t38.state == T38_LOCAL_REINVITE)) {
            change_t38_state(p, T38_ENABLED);
         } else if ((t38action == SDP_T38_INITIATE) &&
               p->owner && p->lastinvite) {
            change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
            /* If fax detection is enabled then send us off to the fax extension */
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_T38)) {
               ast_channel_lock(p->owner);
               if (strcmp(p->owner->exten, "fax")) {
                  const char *target_context = S_OR(p->owner->macrocontext, p->owner->context);
                  ast_channel_unlock(p->owner);
                  if (ast_exists_extension(p->owner, target_context, "fax", 1,
                     S_COR(p->owner->caller.id.number.valid, p->owner->caller.id.number.str, NULL))) {
                     ast_verbose(VERBOSE_PREFIX_2 "Redirecting '%s' to fax extension due to peer T.38 re-INVITE\n", p->owner->name);
                     pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", p->owner->exten);
                     if (ast_async_goto(p->owner, target_context, "fax", 1)) {
                        ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", p->owner->name, target_context);
                     }
                  } else {
                     ast_log(LOG_NOTICE, "T.38 re-INVITE detected but no fax extension\n");
                  }
               } else {
                  ast_channel_unlock(p->owner);
               }
            }
         }
      } else {
         ast_udptl_stop(p->udptl);
         if (debug)
            ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
      }
   }

   if ((portno == -1) && (p->t38.state != T38_DISABLED)) {
      ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
      return 0;
        }

   /* Ok, we're going with this offer */
   ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));

   if (!p->owner)    /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
      return 0;

   ast_debug(4, "We have an owner, now see if we need to change this call\n");

   if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
      if (debug) {
         char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
         ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
            ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
            ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
      }
      p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
      ast_set_read_format(p->owner, p->owner->readformat);
      ast_set_write_format(p->owner, p->owner->writeformat);
   }
   
   if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) {
      ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      /* Queue Manager Unhold event */
      append_history(p, "Unhold", "%s", req->data->str);
      if (sip_cfg.callevents)
         ast_manager_event(p->owner, EVENT_FLAG_CALL, "Hold",
                  "Status: Off\r\n"
                  "Channel: %s\r\n"
                  "Uniqueid: %s\r\n",
                  p->owner->name,
                  p->owner->uniqueid);
      if (sip_cfg.notifyhold)
         sip_peer_hold(p, FALSE);
      ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
   } else if ((ast_sockaddr_isnull(sa) && ast_sockaddr_isnull(vsa) && ast_sockaddr_isnull(tsa) && ast_sockaddr_isnull(isa)) || (sendonly && sendonly != -1)) {
      int already_on_hold = ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD);
      ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
                   S_OR(p->mohsuggest, NULL),
                   !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
      if (sendonly)
         ast_rtp_instance_stop(p->rtp);
      /* RTCP needs to go ahead, even if we're on hold!!! */
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      /* Queue Manager Hold event */
      append_history(p, "Hold", "%s", req->data->str);
      if (sip_cfg.callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
         ast_manager_event(p->owner, EVENT_FLAG_CALL, "Hold",
                  "Status: On\r\n"
                  "Channel: %s\r\n"
                  "Uniqueid: %s\r\n",
                  p->owner->name,
                  p->owner->uniqueid);
      }
      if (sendonly == 1)   /* One directional hold (sendonly/recvonly) */
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
      else if (sendonly == 2) /* Inactive stream */
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
      else
         ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE);
      if (sip_cfg.notifyhold && !already_on_hold)
         sip_peer_hold(p, TRUE);
   }
   
   return 0;
}

static int process_sdp_a_audio	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newaudiortp,
		int *	last_rtpmap_codec
	)			[static]

Definition at line 8738 of file chan_sip.c.

References ast_codec_pref_setsize(), ast_debug, AST_FORMAT_G719, AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, ast_getformatname(), ast_log(), ast_rtp_codecs_packetization_set(), ast_rtp_codecs_payload_lookup(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_payloads_unset(), ast_rtp_instance_get_codecs(), AST_RTP_MAX_PT, AST_RTP_OPT_G726_NONSTANDARD, ast_test_flag, ast_verbose, ast_rtp_payload_type::asterisk_format, ast_rtp_payload_type::code, debug, FALSE, format, LOG_WARNING, LONG_MAX, LONG_MIN, ast_rtp_codecs::pref, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   char fmtp_string[64];
   unsigned int sample_rate;
   int debug = sip_debug_test_pvt(p);

   if (!strncasecmp(a, "ptime", 5)) {
      char *tmp = strrchr(a, ':');
      long int framing = 0;
      if (tmp) {
         tmp++;
         framing = strtol(tmp, NULL, 10);
         if (framing == LONG_MIN || framing == LONG_MAX) {
            framing = 0;
            ast_debug(1, "Can't read framing from SDP: %s\n", a);
         }
      }
      if (framing && p->autoframing) {
         struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
         int codec_n;
         for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
            struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
            if (!format.asterisk_format || !format.code) /* non-codec or not found */
               continue;
            ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(format.code), framing);
            ast_codec_pref_setsize(pref, format.code, framing);
         }
         ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
      }
      found = TRUE;
   } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
             ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -1) {
            if (debug)
               ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
            //found_rtpmap_codecs[last_rtpmap_codec] = codec;
            (*last_rtpmap_codec)++;
            found = TRUE;
         } else {
            ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
            if (debug)
               ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   } else if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
      struct ast_rtp_payload_type payload;

      payload = ast_rtp_codecs_payload_lookup(newaudiortp, codec);
      if (payload.code && payload.asterisk_format) {
         unsigned int bit_rate;

         switch (payload.code) {
         case AST_FORMAT_SIREN7:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 32000) {
                  ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
            break;
         case AST_FORMAT_SIREN14:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 48000) {
                  ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
            break;
         case AST_FORMAT_G719:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 64000) {
                  ast_log(LOG_WARNING, "Got G.719 offer at %d bps, but only 64000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
         }
      }
   }

   return found;
}

static int process_sdp_a_image	(	const char *	a,
		struct sip_pvt *	p
	)			[static]

Definition at line 8917 of file chan_sip.c.

References ast_debug, AST_T38_RATE_12000, AST_T38_RATE_14400, AST_T38_RATE_2400, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600, AST_T38_RATE_MANAGEMENT_LOCAL_TCF, AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF, ast_udptl_set_error_correction_scheme(), ast_udptl_set_far_max_datagram(), FALSE, TRUE, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, and UDPTL_ERROR_CORRECTION_REDUNDANCY.

Referenced by process_sdp().

{
   int found = FALSE;
   char s[256];
   unsigned int x;

   if ((sscanf(a, "T38FaxMaxBuffer:%30u", &x) == 1)) {
      ast_debug(3, "MaxBufferSize:%d\n", x);
      found = TRUE;
   } else if ((sscanf(a, "T38MaxBitRate:%30u", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30u", &x) == 1)) {
      ast_debug(3, "T38MaxBitRate: %d\n", x);
      switch (x) {
      case 14400:
         p->t38.their_parms.rate = AST_T38_RATE_14400;
         break;
      case 12000:
         p->t38.their_parms.rate = AST_T38_RATE_12000;
         break;
      case 9600:
         p->t38.their_parms.rate = AST_T38_RATE_9600;
         break;
      case 7200:
         p->t38.their_parms.rate = AST_T38_RATE_7200;
         break;
      case 4800:
         p->t38.their_parms.rate = AST_T38_RATE_4800;
         break;
      case 2400:
         p->t38.their_parms.rate = AST_T38_RATE_2400;
         break;
      }
      found = TRUE;
   } else if ((sscanf(a, "T38FaxVersion:%30u", &x) == 1)) {
      ast_debug(3, "FaxVersion: %u\n", x);
      p->t38.their_parms.version = x;
      found = TRUE;
   } else if ((sscanf(a, "T38FaxMaxDatagram:%30u", &x) == 1) || (sscanf(a, "T38MaxDatagram:%30u", &x) == 1)) {
      /* override the supplied value if the configuration requests it */
      if (((signed int) p->t38_maxdatagram >= 0) && ((unsigned int) p->t38_maxdatagram > x)) {
         ast_debug(1, "Overriding T38FaxMaxDatagram '%d' with '%d'\n", x, p->t38_maxdatagram);
         x = p->t38_maxdatagram;
      }
      ast_debug(3, "FaxMaxDatagram: %u\n", x);
      ast_udptl_set_far_max_datagram(p->udptl, x);
      found = TRUE;
   } else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
      if (sscanf(a, "T38FaxFillBitRemoval:%30u", &x) == 1) {
         ast_debug(3, "FillBitRemoval: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.fill_bit_removal = TRUE;
         }
      } else {
         ast_debug(3, "FillBitRemoval\n");
         p->t38.their_parms.fill_bit_removal = TRUE;
      }
      found = TRUE;
   } else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
      if (sscanf(a, "T38FaxTranscodingMMR:%30u", &x) == 1) {
         ast_debug(3, "Transcoding MMR: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.transcoding_mmr = TRUE;
         }
      } else {
         ast_debug(3, "Transcoding MMR\n");
         p->t38.their_parms.transcoding_mmr = TRUE;
      }
      found = TRUE;
   } else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
      if (sscanf(a, "T38FaxTranscodingJBIG:%30u", &x) == 1) {
         ast_debug(3, "Transcoding JBIG: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.transcoding_jbig = TRUE;
         }
      } else {
         ast_debug(3, "Transcoding JBIG\n");
         p->t38.their_parms.transcoding_jbig = TRUE;
      }
      found = TRUE;
   } else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
      ast_debug(3, "RateManagement: %s\n", s);
      if (!strcasecmp(s, "localTCF"))
         p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_LOCAL_TCF;
      else if (!strcasecmp(s, "transferredTCF"))
         p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF;
      found = TRUE;
   } else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
      ast_debug(3, "UDP EC: %s\n", s);
      if (!strcasecmp(s, "t38UDPRedundancy")) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
      } else if (!strcasecmp(s, "t38UDPFEC")) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
      } else {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
      }
      found = TRUE;
   }

   return found;
}

static int process_sdp_a_sendonly	(	const char *	a,
		int *	sendonly
	)			[static]

Definition at line 8718 of file chan_sip.c.

References FALSE, and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;

   if (!strcasecmp(a, "sendonly")) {
      if (*sendonly == -1)
         *sendonly = 1;
      found = TRUE;
   } else if (!strcasecmp(a, "inactive")) {
      if (*sendonly == -1)
         *sendonly = 2;
      found = TRUE;
   }  else if (!strcasecmp(a, "sendrecv")) {
      if (*sendonly == -1)
         *sendonly = 0;
      found = TRUE;
   }
   return found;
}

static int process_sdp_a_text	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newtextrtp,
		char *	red_fmtp,
		int *	red_num_gen,
		int *	red_data_pt,
		int *	last_rtpmap_codec
	)			[static]

Definition at line 8868 of file chan_sip.c.

References AST_RED_MAX_GENERATION, ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_verbose, debug, FALSE, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   unsigned int sample_rate;
   char *red_cp;
   int debug = sip_debug_test_pvt(p);

   if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
            if (p->trtp) {
               /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
               ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
               found = TRUE;
            }
         } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
            if (p->trtp) {
               ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
               sprintf(red_fmtp, "fmtp:%d ", codec);
               if (debug)
                  ast_verbose("RED submimetype has payload type: %d\n", codec);
               found = TRUE;
            }
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
      /* count numbers of generations in fmtp */
      red_cp = &red_fmtp[strlen(red_fmtp)];
      strncpy(red_fmtp, a, 100);

      sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
      red_cp = strtok(red_cp, "/");
      while (red_cp && (*red_num_gen)++ < AST_RED_MAX_GENERATION) {
         sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
         red_cp = strtok(NULL, "/");
      }
      red_cp = red_fmtp;
      found = TRUE;
   }

   return found;
}

static int process_sdp_a_video	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newvideortp,
		int *	last_rtpmap_codec
	)			[static]

Definition at line 8834 of file chan_sip.c.

References ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_payloads_unset(), ast_verbose, debug, FALSE, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   unsigned int sample_rate;
   int debug = sip_debug_test_pvt(p);

   if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         /* Note: should really look at the '#chans' params too */
         if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
            if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
               if (debug)
                  ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
               //found_rtpmap_codecs[last_rtpmap_codec] = codec;
               (*last_rtpmap_codec)++;
               found = TRUE;
            } else {
               ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
               if (debug)
                  ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
            }
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   }

   return found;
}

static int process_sdp_c	(	const char *	c,
		struct ast_sockaddr *	addr
	)			[static]

Definition at line 8691 of file chan_sip.c.

References ast_log(), ast_sockaddr_resolve_first_af(), FALSE, LOG_WARNING, and TRUE.

Referenced by process_sdp().

{
   char proto[4], host[258];
   int af;

   /* Check for Media-description-level-address */
   if (sscanf(c, "IN %3s %255s", proto, host) == 2) {
      if (!strcmp("IP4", proto)) {
         af = AF_INET;
      } else if (!strcmp("IP6", proto)) {
         af = AF_INET6;
      } else {
         ast_log(LOG_WARNING, "Unknown protocol '%s'.\n", proto);
         return FALSE;
      }
      if (ast_sockaddr_resolve_first_af(addr, host, 0, af)) {
         ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in c= line, '%s'\n", c);
         return FALSE;
      }
      return TRUE;
   } else {
      ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
      return FALSE;
   }
   return FALSE;
}

static int process_sdp_o	(	const char *	o,
		struct sip_pvt *	p
	)			[static]

Definition at line 8612 of file chan_sip.c.

References ast_debug, ast_log(), ast_strdupa, ast_strlen_zero(), ast_test_flag, FALSE, LOG_WARNING, strsep(), and TRUE.

Referenced by process_sdp().

{
   char *o_copy;
   char *token;
   int64_t rua_version;

   /* Store the SDP version number of remote UA. This will allow us to
   distinguish between session modifications and session refreshes. If
   the remote UA does not send an incremented SDP version number in a
   subsequent RE-INVITE then that means its not changing media session.
   The RE-INVITE may have been sent to update connected party, remote
   target or to refresh the session (Session-Timers).  Asterisk must not
   change media session and increment its own version number in answer
   SDP in this case. */

   p->session_modify = TRUE;

   if (ast_strlen_zero(o)) {
      ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
      return FALSE;
   }

   o_copy = ast_strdupa(o);
   token = strsep(&o_copy, " ");  /* Skip username   */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
      return FALSE;
   }
   token = strsep(&o_copy, " ");  /* Skip session-id */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line session-id\n");
      return FALSE;
   }
   token = strsep(&o_copy, " ");  /* Version         */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line\n");
      return FALSE;
   }
   if (!sscanf(token, "%30" SCNd64, &rua_version)) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line version\n");
      return FALSE;
   }

   /* we need to check the SDP version number the other end sent us;
    * our rules for deciding what to accept are a bit complex.
    *
    * 1) if 'ignoresdpversion' has been set for this dialog, then
    *    we will just accept whatever they sent and assume it is
    *    a modification of the session, even if it is not
    * 2) otherwise, if this is the first SDP we've seen from them
    *    we accept it
    * 3) otherwise, if the new SDP version number is higher than the
    *    old one, we accept it
    * 4) otherwise, if this SDP is in response to us requesting a switch
    *    to T.38, we accept the SDP, but also generate a warning message
    *    that this peer should have the 'ignoresdpversion' option set,
    *    because it is not following the SDP offer/answer RFC; if we did
    *    not request a switch to T.38, then we stop parsing the SDP, as it
    *    has not changed from the previous version
    */

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
       (p->sessionversion_remote < 0) ||
       (p->sessionversion_remote < rua_version)) {
      p->sessionversion_remote = rua_version;
   } else {
      if (p->t38.state == T38_LOCAL_REINVITE) {
         p->sessionversion_remote = rua_version;
         ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
      } else {
         p->session_modify = FALSE;
         ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
         return FALSE;
      }
   }

   return TRUE;
}

static int proxy_update

(

struct sip_proxy *

proxy

)

[static]

Resolve DNS srv name or host name in a sip_proxy structure

Definition at line 2838 of file chan_sip.c.

References ast_get_ip_or_srv(), ast_log(), ast_sockaddr_parse(), ast_sockaddr_set_port, bindaddr, FALSE, get_address_family_filter(), LOG_WARNING, sip_cfg, and TRUE.

Referenced by build_peer(), and reload_config().

{
   /* if it's actually an IP address and not a name,
           there's no need for a managed lookup */
   if (!ast_sockaddr_parse(&proxy->ip, proxy->name, 0)) {
      /* Ok, not an IP address, then let's check if it's a domain or host */
      /* XXX Todo - if we have proxy port, don't do SRV */
      proxy->ip.ss.ss_family = get_address_family_filter(&bindaddr); /* Filter address family */
      if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
            ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
            return FALSE;
      }

   }

   ast_sockaddr_set_port(&proxy->ip, proxy->port);

   proxy->last_dnsupdate = time(NULL);
   return TRUE;
}

static int publish_expire

(

const void *

data

)

[static]

Definition at line 1001 of file chan_sip.c.

References ao2_ref, ao2_unlink, ast_assert, event_state_compositor::compositor, and get_esc().

Referenced by create_esc_entry(), handle_sip_publish_modify(), and handle_sip_publish_refresh().

{
   struct sip_esc_entry *esc_entry = (struct sip_esc_entry *) data;
   struct event_state_compositor *esc = get_esc(esc_entry->event);

   ast_assert(esc != NULL);

   ao2_unlink(esc->compositor, esc_entry);
   ao2_ref(esc_entry, -1);
   return 0;
}

static void pvt_set_needdestroy	(	struct sip_pvt *	pvt,
		const char *	reason
	)			[inline, static]

Definition at line 2803 of file chan_sip.c.

References append_history.

Referenced by __sip_autodestruct(), handle_incoming(), handle_request_refer(), handle_request_subscribe(), handle_response(), handle_response_invite(), handle_response_notify(), handle_response_peerpoke(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), retrans_pkt(), sip_hangup(), and sip_reg_timeout().

{
   if (pvt->final_destruction_scheduled) {
      return; /* This is already scheduled for final destruction, let the scheduler take care of it. */
   }
   append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
   pvt->needdestroy = 1;
}

static int queue_request	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)			[static]

Definition at line 23711 of file chan_sip.c.

References ast_calloc, AST_LIST_INSERT_TAIL, ast_sched_add(), copy_request(), and scheduler_process_request_queue().

Referenced by handle_request_do().

{
   struct sip_request *newreq;

   if (!(newreq = ast_calloc(1, sizeof(*newreq)))) {
      return -1;
   }

   copy_request(newreq, req);
   AST_LIST_INSERT_TAIL(&p->request_queue, newreq, next);
   if (p->request_queue_sched_id == -1) {
      if ((p->request_queue_sched_id = ast_sched_add(sched, 10, scheduler_process_request_queue, dialog_ref(p, "Increment refcount to pass dialog pointer to sched callback"))) == -1) {
         dialog_unref(p, "Decrement refcount due to sched_add failure");
      }
   }

   return 0;
}

static struct sip_peer * realtime_peer	(	const char *	peername,
		struct ast_sockaddr *	sin,
		int	devstate_only
	)			[static, read]

realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.

Note:

If this one loaded something, then we need to ensure that the host field matched. The only reason why we can't have this as a criteria is because we only have the IP address and the host field might be set as a name (and the reverse PTR might not match).

Definition at line 4374 of file chan_sip.c.

References ao2_t_link, ast_check_realtime(), ast_config_destroy(), ast_copy_flags, ast_copy_string(), ast_debug, ast_free, ast_load_realtime(), ast_load_realtime_multientry(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_sockaddr_cmp(), ast_sockaddr_isnull(), ast_sockaddr_resolve(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), ast_test_flag, ast_variables_destroy(), bindaddr, build_peer(), expire_register(), get_address_family_filter(), get_insecure_variable_from_config(), get_name_from_variable(), ipaddr, LOG_WARNING, ast_variable::name, ast_variable::next, PARSE_PORT_FORBID, ref_peer(), SENTINEL, sip_cfg, TRUE, unref_peer(), ast_variable::value, and var.

{
   struct sip_peer *peer;
   struct ast_variable *var = NULL;
   struct ast_variable *varregs = NULL;
   struct ast_variable *tmp;
   struct ast_config *peerlist = NULL;
   char ipaddr[INET_ADDRSTRLEN];
   char portstring[6]; /*up to 5 digits plus null terminator*/
   char *cat = NULL;
   int realtimeregs = ast_check_realtime("sipregs");

   /* First check on peer name */
   if (newpeername) {
      if (realtimeregs)
         varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);

      var = ast_load_realtime("sippeers", "name", newpeername, "host", "dynamic", SENTINEL);
      if (!var && addr) {
         var = ast_load_realtime("sippeers", "name", newpeername, "host", ast_sockaddr_stringify_addr(addr), SENTINEL);
      }
      if (!var) {
         var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
         /*!\note
          * If this one loaded something, then we need to ensure that the host
          * field matched.  The only reason why we can't have this as a criteria
          * is because we only have the IP address and the host field might be
          * set as a name (and the reverse PTR might not match).
          */
         if (var && addr) {
            for (tmp = var; tmp; tmp = tmp->next) {
               if (!strcasecmp(tmp->name, "host")) {
                  struct ast_sockaddr *addrs = NULL;

                  if (ast_sockaddr_resolve(&addrs,
                            tmp->value,
                            PARSE_PORT_FORBID,
                            get_address_family_filter(&bindaddr)) <= 0 ||
                      ast_sockaddr_cmp(&addrs[0], addr)) {
                     /* No match */
                     ast_variables_destroy(var);
                     var = NULL;
                  }
                  ast_free(addrs);
                  break;
               }
            }
         }
      }
   }

   if (!var && addr) {  /* Then check on IP address for dynamic peers */
      ast_copy_string(ipaddr, ast_sockaddr_stringify_addr(addr), sizeof(ipaddr));
      ast_copy_string(portstring, ast_sockaddr_stringify_port(addr), sizeof(portstring));
      var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, SENTINEL);  /* First check for fixed IP hosts */
      if (var) {
         if (realtimeregs) {
            newpeername = get_name_from_variable(var, newpeername);
            varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
         }
      } else {
         if (realtimeregs)
            varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
         else
            var = ast_load_realtime("sippeers", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
         if (varregs) {
            newpeername = get_name_from_variable(varregs, newpeername);
            var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
         }
      }
      if (!var) { /*We couldn't match on ipaddress and port, so we need to check if port is insecure*/
         peerlist = ast_load_realtime_multientry("sippeers", "host", ipaddr, SENTINEL);
         if (peerlist) {
            var = get_insecure_variable_from_config(peerlist);
            if(var) {
               if (realtimeregs) {
                  newpeername = get_name_from_variable(var, newpeername);
                  varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
               }
            } else { /*var wasn't found in the list of "hosts", so try "ipaddr"*/
               peerlist = NULL;
               cat = NULL;
               peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
               if(peerlist) {
                  var = get_insecure_variable_from_config(peerlist);
                  if(var) {
                     if (realtimeregs) {
                        newpeername = get_name_from_variable(var, newpeername);
                        varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
                     }
                  }
               }
            }
         } else {
            if (realtimeregs) {
               peerlist = ast_load_realtime_multientry("sipregs", "ipaddr", ipaddr, SENTINEL);
               if (peerlist) {
                  varregs = get_insecure_variable_from_config(peerlist);
                  if (varregs) {
                     newpeername = get_name_from_variable(varregs, newpeername);
                     var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
                  }
               }
            } else {
               peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
               if (peerlist) {
                  var = get_insecure_variable_from_config(peerlist);
                  if (var) {
                     newpeername = get_name_from_variable(var, newpeername);
                     varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
                  }
               }
            }
         }
      }
   }

   if (!var) {
      if (peerlist)
         ast_config_destroy(peerlist);
      return NULL;
   }

   for (tmp = var; tmp; tmp = tmp->next) {
      if (!newpeername && !strcasecmp(tmp->name, "name")) {
         newpeername = tmp->value;
      }
   }
   
   if (!newpeername) {  /* Did not find peer in realtime */
      ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
      if(peerlist)
         ast_config_destroy(peerlist);
      else
         ast_variables_destroy(var);
      return NULL;
   }


   /* Peer found in realtime, now build it in memory */
   peer = build_peer(newpeername, var, varregs, TRUE, devstate_only);
   if (!peer) {
      if(peerlist)
         ast_config_destroy(peerlist);
      else {
         ast_variables_destroy(var);
         ast_variables_destroy(varregs);
      }
      return NULL;
   }

   ast_debug(3, "-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);

   if (ast_test_flag(&