The Asterisk Development Team would like to announce the first release candidate of Asterisk 15.1.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Improvements made in this release:
———————————–
|
chan_sip: Provide access to read the full SIP Request-URI from INVITE (Reported by David J. Pryke) |
|
|
alembic: Add support for Microsoft SQL server (Reported by Florian Floimair) |
|
|
libsrtp-2.1.x support (Reported by Alexander Traud) |
|
|
Enable CHANNEL function to get from and to tag from SIP Headers (Reported by Andre Nazario) |
|
|
Google OAuth 2.0 support for XMPP / Motif (Reported by Andrey) |
|
|
Support for GMIME 3.0 (Reported by Tzafrir Cohen) |
|
|
chan_pjsip: Port SIPDtmfMode to chan_pjsip (Reported by Torrey Searle) |
Bugs fixed in this release:
———————————–
|
res_pjsip: user=phone added to Anonymous caller-id when it shouldn’t be. (Reported by dtryba) |
|
|
res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs (Reported by dtryba) |
|
|
cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured (Reported by Tzafrir Cohen) |
|
|
Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate (Reported by Allen Ford) |
|
|
res_pjsip: Loss of SIP registrations causing unavailable endpoints (Reported by Richard Mudgett) |
|
|
res_ari: Memory leaks in ARI when using Content-Type: application/json (Reported by David Hajek) |
|
|
chan_sip: tcpbind uses wrong source address (Reported by Ksenia) |
|
|
Dual-Stack server cannot be used as IPv4 client via TCP/TLS (Reported by Alexander Traud) |
|
|
vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell) |
|
|
app_queue: Music On Hold for real-time queues is not reset to default (Reported by Nathan Bruning) |
|
|
res_pjsip_mwi: uninitialized value from ast_strings_match (Reported by Corey Farrell) |
|
|
Status of RFC 3323 and PJSIP (Reported by dtryba) |
|
|
False positive busy checks when icalendar’s recurrence-id mechanism is involved (Reported by Benoît Dereck-Tricot) |
|
|
app_queue: does its check-makeannouncement-logic twice each head-caller-loop (Reported by Stefan Engström) |
|
|
Problem with expires on pjsip / outbound-publish (Reported by Cyrille Demaret) |
|
|
Contact is improperly translated after d178f497 (Reported by Sean Bright) |
|
|
Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) (Reported by Ross Beer) |
|
|
chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller (Reported by lvl) |
|
|
A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) |
|
|
bridge: Renegotiate if source stream changes. (Reported by Joshua Colp) |
|
|
Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) |
|
|
res_pjsip_session: Crashes after sending PRACK and receiving 200 OK (Reported by Daniel Heckl) |
|
|
[pjsip] chan_pjsip_indicate: Don’t know how to indicate condition 36 (Reported by Daniel Heckl) |
|
|
bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis – Prescom) |
|
|
SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 (Reported by Marcello Ceschia) |
|
|
RTCP needs better packet validation to resist port scans. (Reported by Richard Mudgett) |
|
|
RTP: One way audio with direct media and strictrtp=yes. (Reported by Richard Mudgett) |
|
|
Crash in pubsub_on_rx_request NULL pointer – Possible PJSIP Vulnerability (Reported by Ross Beer) |
|
|
module reload res_calendar.so does not reload everything in calendar.conf (Reported by Jesper) |
|
|
res_calendar does not process CalDAV from Owncloud [fix included] (Reported by Stefan Gofferje) |
|
|
res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. (Reported by Jesper) |
|
|
RTP Multicast of L16 (type 10): Asterisk and wireshark disagree (Reported by Tzafrir Cohen) |
|
|
external_media_address and external_signaling_address don’t always honor localnet (Reported by Walter Doekes) |
|
|
res_smdi: convert to astobj2 (Reported by Corey Farrell) |
|
|
chan_sip: Asterisk crashing when subscription doesn’t get set (Reported by Bryan Walters) |
|
|
SDP origin attribute modified when issuing re-INVITE because of directmedia=yes (Reported by saghul) |
|
|
CDR: CDR(start,u) function won’t work in cdr_custom config (Reported by Jacek Konieczny) |
|
|
alembic: prune_on_boot fix erroneous (Reported by Florian Floimair) |
|
|
When in queue on g722 with interruptions, music on hold can get stuck and no longer play (Reported by Jens T.) |
|
|
nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) |
|
|
PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) |
|
|
Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) |
|
|
Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) |
|
|
ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen) |
|
|
libc segfault upon entry into app_directory (Reported by David Moore) |
|
|
Sending a “tel” uri in a From or To header in an unauthenticated message causes asterisk to crash (Reported by Ross Beer) |
|
|
core: ast_safe_system command injection possible. (Reported by Corey Farrell) |
|
|
res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) |
|
|
res_rtp_asterisk: Allow remote SSRC to change due to renegotiation (Reported by Joshua Colp) |
|
|
Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name (Reported by James Terhune) |
|
|
core: Don’t queue up multiple video update frames. (Reported by Joshua Colp) |
|
|
app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) |
|
|
several filename bugs in Record() application (Reported by klaus3000) |
|
|
alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) |
|
|
Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) |
|
|
When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. (Reported by Jim Van Meggelen) |
|
|
When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored (Reported by Eelco Brolman) |
|
|
bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU (Reported by Richard Mudgett) |
|
|
app_queue: Wrong queue stat calculation (Reported by sungtae kim) |
|
|
XMPP OAuth not working due to inverted logic (Reported by Michael Kuron) |
|
|
res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical (Reported by Mark Thompson) |
|
|
If wget is not installed and “or” is not available, external components (excluding pjsip) are not installed (Reported by Seán C. McCord) |
|
|
manager: hook event is not being raised (Reported by Kevin Harwell) |
|
|
Either asterisk or pjproject isn’t re-using tcp connections (again) (Reported by George Joseph) |
|
|
IPv6 receive address in message doesn’t include brackets (Reported by Scott Griepentrog) |
|
|
res_rtp_asterisk: RTCP statistics are not available when native bridge is used (Reported by Torrey Searle) |
|
|
Asymmetric codecs when asymmetric_rtp_codec=no (Reported by Jesse Ross) |
|
|
Make –with-pjproject-bundled the default for Asterisk 15 (Reported by George Joseph) |
|
|
RTP session is not fully destroyed on channel hangup (Reported by Matt Jordan) |
|
|
bridge: Crash when mapping streams (Reported by Joshua Colp) |
|
|
channel: requester leaks joint_cap on success. (Reported by Corey Farrell) |
|
|
res_pjsip_session: Handling of ‘msid’ is incorrect (Reported by Kevin Harwell) |
|
|
res_pjsip: parse/add msid attribute when webrtc is enabled (Reported by Kevin Harwell) |
|
|
Asterisk 15.0.0-Beta1 does not compile (Reported by Ira Emus) |
|
|
res_pjsip: PJSIP presence – missing braces around the status element in XML (Reported by Abraham Liebsch) |
|
|
Asterisk won’t compile on Fedora 26 with devmode enabled. (Reported by Corey Farrell) |
|
|
res_pjsip: TLS connection not stable (Reported by Ian Gilmour) |
New Features made in this release:
———————————–
|
AMI : Add CancelAtxfer Action (Reported by Thomas Sevestre) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0-rc1
Thank you for your continued support of Asterisk!