Asterisk News

Asterisk Releases

Asterisk 13.3.0-rc1 Now Available

Mar 23, 2015

The Asterisk Development Team has announced the first release candidate of Asterisk 13.3.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22670] - Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24015] - app_transfer fails with PJSIP channels
  • [ASTERISK-24085] - Documentation - We should remove or further document the 'contact' section in pjsip.conf
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24612] - res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24616] - Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24632] - install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24677] - ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24685] - "pjsip show version" CLI command
  • [ASTERISK-24689] - Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24700] - CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24716] - Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24727] - PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] - [patch]Segmentation fault on aoc-e event
  • [ASTERISK-24741] - dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24751] - Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] - res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] - res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] - ${CHANNEL(pjsip)} - segfault
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24785] - 'Expires' header missing from 200 OK on REGISTER
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24791] - Crash in ast_rtcp_write_report
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24807] - Missing mandatory field Max-Forwards
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24830] - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] - res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24872] - [patch] AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD
  • [ASTERISK-24882] - chan_sip: Improve usage of REF_DEBUG

Improvement

  • [ASTERISK-24745] - [patch]Add no_answer to ARI hangup causes
  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context
  • [ASTERISK-24811] - asterisk-publication sorcery object does not use realtime

New Feature

  • [ASTERISK-17899] - [patch] Adds a 'ignorecryptolifetime' (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional 'lifetime'
  • [ASTERISK-24703] - ARI: Add the ability to "transfer" (redirect) a channel


For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0-rc1


Thank you for your continued support of Asterisk!


Asterisk 11.17.0-rc1 Now Available

Mar 23, 2015

The Asterisk Development Team has announced the first release candidate of Asterisk 11.17.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.17.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22436] - [patch] No BYE to masqueraded channel on INVITE with replaces
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD

Improvement

  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context

New Feature

  • [ASTERISK-17899] - [patch] Adds a 'ignorecryptolifetime' (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional 'lifetime'

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 13.2.0 Now Available

Feb 6, 2015

The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.2.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Improvements

  • [ASTERISK-24316] - For httpd server, need option to define server name for security purposes
  • [ASTERISK-24412] - [patch]Incomplete channel originate/continue handling with ARI
  • [ASTERISK-24552] - ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
  • [ASTERISK-24553] - ARI/AMI: Include language in standard channel snapshot output
  • [ASTERISK-24575] - [patch]Make capath work for res_pjsip
  • [ASTERISK-24643] - res_pjsip: Add user=phone option
  • [ASTERISK-24644] - res_pjsip_keepalive: Add keepalive module for connection-oriented transports.
  • [ASTERISK-24671] - Missing docs for the CDR AMI Event
  • [ASTERISK-24678] - [PATCH] Added atxfer* settings to features.conf.sample

Bugs

  • [ASTERISK-20744] - [patch] Security event logging does not work over syslog
  • [ASTERISK-23733] - 'reload acl' fails if acl.conf is not present on startup
  • [ASTERISK-23841] - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.
  • [ASTERISK-23850] - Park Application does not respect Return Context Priority
  • [ASTERISK-23991] - [patch]asterisk.pc file contains a small error in the CFlags returned
  • [ASTERISK-24048] - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts
  • [ASTERISK-24049] - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack
  • [ASTERISK-24288] - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
  • [ASTERISK-24337] - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'
  • [ASTERISK-24342] - PJSIP: Qualifying endpoints attempts to do them all at the same time.
  • [ASTERISK-24355] - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'
  • [ASTERISK-24376] - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI
  • [ASTERISK-24449] - Reinvite for T.38 UDPTL fails if SRTP is enabled
  • [ASTERISK-24459] - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible
  • [ASTERISK-24472] - Asterisk Crash in OpenSSL when calling over WSS from JSSIP
  • [ASTERISK-24474] - sip_to_pjsip.py lacks documentation and does not function
  • [ASTERISK-24485] - res_pjsip cannot be unloaded or shutdown
  • [ASTERISK-24513] - Local channel apparently leaked in off-nominal DTMF attended transfer
  • [ASTERISK-24514] - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard
  • [ASTERISK-24536] - AMI redirect with PJSIP fails to move extra channel
  • [ASTERISK-24539] - Compile fails on OSX because of sem_timedwait in bridge_channel.c
  • [ASTERISK-24544] - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll
  • [ASTERISK-24560] - Creating a named ARI bridge twice causes a crash
  • [ASTERISK-24563] - Direct Media calls within private network sometimes get one way audio
  • [ASTERISK-24591] - Stasis() side of an ARI originated channel cannot be Redirected
  • [ASTERISK-24600] - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock
  • [ASTERISK-24604] - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core
  • [ASTERISK-24607] - res_pjsip_session: re-INVITE with declined media streams results in 488
  • [ASTERISK-24614] - Deadlock when DEBUG_THREADS compiler flag enabled
  • [ASTERISK-24615] - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE
  • [ASTERISK-24619] - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
  • [ASTERISK-24624] - Transfer to invalid extension results in hung channel.
  • [ASTERISK-24626] - Voicemail passwords not being stored in ARA
  • [ASTERISK-24628] - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
  • [ASTERISK-24635] - PJSIP outbound PUBLISH crashes when no response is ever received
  • [ASTERISK-24637] - Channel re-enters Stasis() when it should not
  • [ASTERISK-24640] - Registration pending stays forever after sip reload
  • [ASTERISK-24646] - PJSIP changeset 4899 breaks TLS
  • [ASTERISK-24649] - Pushing of channel into bridge fails; Stasis fails to get app name
  • [ASTERISK-24655] - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
  • [ASTERISK-24663] - [patch] Unnamed semaphore autoconf check fails on cross compilation
  • [ASTERISK-24665] - Configure check required for pjsip_get_dest_info()
  • [ASTERISK-24666] - Security Vulnerability: RTP not closed after sip call using unsupported codec
  • [ASTERISK-24672] - [PATCH] Memory leak in func_curl CURLOPT
  • [ASTERISK-24673] - outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)
  • [ASTERISK-24676] - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
  • [ASTERISK-24682] - app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value
  • [ASTERISK-24693] - Investigate and fix memory leaks in Asterisk
  • [ASTERISK-24709] - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event
  • [ASTERISK-24711] - DTLS handshake broken with latest OpenSSL versions
  • [ASTERISK-24715] - chan_sip: stale nonce causes failure
  • [ASTERISK-24719] - ConfBridge recording channels get stuck when recording started/stopped more than once
  • [ASTERISK-24721] - manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation
  • [ASTERISK-24723] - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus
  • [ASTERISK-24728] - tcptls: Bad file descriptor error when reloading chan_sip
  • [ASTERISK-24729] - Outbound registration not occuring on new registrations after reload.
  • [ASTERISK-24736] - Memory Leak Fixes
  • [ASTERISK-24737] - When agent not logged in, agent status shows unavailable, queue status shows agent invalid

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0

Thank you for your continued support of Asterisk!


Asterisk 11.16.0 Now Available

Feb 6, 2015

The Asterisk Development Team has announced the release of Asterisk 11.16.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.16.0 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Bugs

  • [ASTERISK-20744] - [patch] Security event logging does not work over syslog
  • [ASTERISK-23733] - 'reload acl' fails if acl.conf is not present on startup
  • [ASTERISK-23850] - Park Application does not respect Return Context Priority
  • [ASTERISK-23991] - [patch]asterisk.pc file contains a small error in the CFlags returned
  • [ASTERISK-24048] - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts
  • [ASTERISK-24288] - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
  • [ASTERISK-24337] - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'
  • [ASTERISK-24355] - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'
  • [ASTERISK-24449] - Reinvite for T.38 UDPTL fails if SRTP is enabled
  • [ASTERISK-24472] - Asterisk Crash in OpenSSL when calling over WSS from JSSIP
  • [ASTERISK-24614] - Deadlock when DEBUG_THREADS compiler flag enabled
  • [ASTERISK-24619] - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
  • [ASTERISK-24628] - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
  • [ASTERISK-24646] - PJSIP changeset 4899 breaks TLS
  • [ASTERISK-24672] - [PATCH] Memory leak in func_curl CURLOPT
  • [ASTERISK-24676] - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
  • [ASTERISK-24709] - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event
  • [ASTERISK-24711] - DTLS handshake broken with latest OpenSSL versions
  • [ASTERISK-24715] - chan_sip: stale nonce causes failure
  • [ASTERISK-24719] - ConfBridge recording channels get stuck when recording started/stopped more than once
  • [ASTERISK-24728] - tcptls: Bad file descriptor error when reloading chan_sip

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0

Thank you for your continued support of Asterisk!


Asterisk 13.2.0-rc1 Now Available

Jan 30, 2015

The Asterisk Development Team has announced the release of Asterisk 13.2.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.2.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release:

Improvements

  • [ASTERISK-24316] - For httpd server, need option to define server name for security purposes
  • [ASTERISK-24412] - [patch]Incomplete channel originate/continue handling with ARI
  • [ASTERISK-24552] - ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
  • [ASTERISK-24553] - ARI/AMI: Include language in standard channel snapshot output
  • [ASTERISK-24575] - [patch]Make capath work for res_pjsip
  • [ASTERISK-24643] - res_pjsip: Add user=phone option
  • [ASTERISK-24644] - res_pjsip_keepalive: Add keepalive module for connection-oriented transports.
  • [ASTERISK-24671] - Missing docs for the CDR AMI Event
  • [ASTERISK-24678] - [PATCH] Added atxfer* settings to features.conf.sample

Bugs

  • [ASTERISK-20744] - [patch] Security event logging does not work over syslog
  • [ASTERISK-23733] - 'reload acl' fails if acl.conf is not present on startup
  • [ASTERISK-23841] - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.
  • [ASTERISK-23850] - Park Application does not respect Return Context Priority
  • [ASTERISK-23991] - [patch]asterisk.pc file contains a small error in the CFlags returned
  • [ASTERISK-24048] - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts
  • [ASTERISK-24049] - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack
  • [ASTERISK-24288] - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
  • [ASTERISK-24337] - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'
  • [ASTERISK-24342] - PJSIP: Qualifying endpoints attempts to do them all at the same time.
  • [ASTERISK-24355] - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'
  • [ASTERISK-24376] - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI
  • [ASTERISK-24449] - Reinvite for T.38 UDPTL fails if SRTP is enabled
  • [ASTERISK-24459] - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible
  • [ASTERISK-24472] - Asterisk Crash in OpenSSL when calling over WSS from JSSIP
  • [ASTERISK-24474] - sip_to_pjsip.py lacks documentation and does not function
  • [ASTERISK-24485] - res_pjsip cannot be unloaded or shutdown
  • [ASTERISK-24513] - Local channel apparently leaked in off-nominal DTMF attended transfer
  • [ASTERISK-24514] - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard
  • [ASTERISK-24536] - AMI redirect with PJSIP fails to move extra channel
  • [ASTERISK-24539] - Compile fails on OSX because of sem_timedwait in bridge_channel.c
  • [ASTERISK-24544] - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll
  • [ASTERISK-24560] - Creating a named ARI bridge twice causes a crash
  • [ASTERISK-24563] - Direct Media calls within private network sometimes get one way audio
  • [ASTERISK-24591] - Stasis() side of an ARI originated channel cannot be Redirected
  • [ASTERISK-24600] - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock
  • [ASTERISK-24604] - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core
  • [ASTERISK-24607] - res_pjsip_session: re-INVITE with declined media streams results in 488
  • [ASTERISK-24614] - Deadlock when DEBUG_THREADS compiler flag enabled
  • [ASTERISK-24615] - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE
  • [ASTERISK-24619] - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
  • [ASTERISK-24624] - Transfer to invalid extension results in hung channel.
  • [ASTERISK-24626] - Voicemail passwords not being stored in ARA
  • [ASTERISK-24628] - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
  • [ASTERISK-24635] - PJSIP outbound PUBLISH crashes when no response is ever received
  • [ASTERISK-24637] - Channel re-enters Stasis() when it should not
  • [ASTERISK-24640] - Registration pending stays forever after sip reload
  • [ASTERISK-24646] - PJSIP changeset 4899 breaks TLS
  • [ASTERISK-24649] - Pushing of channel into bridge fails; Stasis fails to get app name
  • [ASTERISK-24655] - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
  • [ASTERISK-24663] - [patch] Unnamed semaphore autoconf check fails on cross compilation
  • [ASTERISK-24665] - Configure check required for pjsip_get_dest_info()
  • [ASTERISK-24666] - Security Vulnerability: RTP not closed after sip call using unsupported codec
  • [ASTERISK-24672] - [PATCH] Memory leak in func_curl CURLOPT
  • [ASTERISK-24673] - outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)
  • [ASTERISK-24676] - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
  • [ASTERISK-24682] - app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value
  • [ASTERISK-24693] - Investigate and fix memory leaks in Asterisk
  • [ASTERISK-24709] - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event
  • [ASTERISK-24711] - DTLS handshake broken with latest OpenSSL versions
  • [ASTERISK-24715] - chan_sip: stale nonce causes failure
  • [ASTERISK-24719] - ConfBridge recording channels get stuck when recording started/stopped more than once
  • [ASTERISK-24721] - manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation
  • [ASTERISK-24723] - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus
  • [ASTERISK-24728] - tcptls: Bad file descriptor error when reloading chan_sip
  • [ASTERISK-24729] - Outbound registration not occuring on new registrations after reload.
  • [ASTERISK-24736] - Memory Leak Fixes
  • [ASTERISK-24737] - When agent not logged in, agent status shows unavailable, queue status shows agent invalid

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0-rc1

Thank you for your continued support of Asterisk!


Pages

Subscribe to