<html><head></head><body>
The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.<br>
This release is available for immediate download at<br>
<a href='http://downloads.asterisk.org/pub/telephony/asterisk'>http://downloads.asterisk.org/pub/telephony/asterisk</a>
<p>
The release of Asterisk 15.0.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
<p>
<b>Thank you!</b><br>
<p>
The following issues are resolved in this release:<br>
<p>
<b>Improvements made in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26230'>ASTERISK-26230</a>] - <td><td>[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27043'>ASTERISK-27043</a>] - <td><td>Core/BuildSystem: Add defines to fix build with LibreSSL<br>(Reported by Guido Falsi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27042'>ASTERISK-27042</a>] - <td><td>Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br>(Reported by Guido Falsi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26419'>ASTERISK-26419</a>] - <td><td>audiohooks: Remove redundant codec translations when using audiohooks<br>(Reported by Michael Walton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26976'>ASTERISK-26976</a>] - <td><td>libsrtp-2.x.x support<br>(Reported by Alex)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27014'>ASTERISK-27014</a>] - <td><td>configurable busy_timeout in sqlite backends<br>(Reported by Marek Cervenka)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26124'>ASTERISK-26124</a>] - <td><td>res_agi: Set audio format for EAGI audio stream<br>(Reported by John Fawcett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26088'>ASTERISK-26088</a>] - <td><td>Investigate heavy memory utilization by res_pjsip_pubsub<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26427'>ASTERISK-26427</a>] - <td><td>res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip<br>(Reported by Nir Simionovich (GreenfieldTech - Israel))</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26932'>ASTERISK-26932</a>] - <td><td>[patch] SIP/SDP: No rtpmap for static RTP payload IDs<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26864'>ASTERISK-26864</a>] - <td><td>res_pjsip_session: Add support for overlap dialling<br>(Reported by Richard Begg)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26846'>ASTERISK-26846</a>] - <td><td>chan_sip: Add rtcp-mux support<br>(Reported by Sean Bright)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26568'>ASTERISK-26568</a>] - <td><td>pbx_spool: OUTGOING_RETRY variable<br>(Reported by Roman Shubovich)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26292'>ASTERISK-26292</a>] - <td><td>app_confbridge: 3D-Conferencing via Binaural Synthesis<br>(Reported by Dennis Guse)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23828'>ASTERISK-23828</a>] - <td><td>pjsip - Need a command to list active SIP subscriptions<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26559'>ASTERISK-26559</a>] - <td><td>app_queue: New service level calculation<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26658'>ASTERISK-26658</a>] - <td><td>Add ability for dialplan show to display filenames/line numbers of registered extensions<br>(Reported by Jonathan R. Rose)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26527'>ASTERISK-26527</a>] - <td><td>Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec<br>(Reported by Badalian Vyacheslav)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22992'>ASTERISK-22992</a>] - <td><td>[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel<br>(Reported by Anthony Messina)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26624'>ASTERISK-26624</a>] - <td><td>res_calendar_caldav: Add support for gmail<br>(Reported by Eduardo Scudeller Libardi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26562'>ASTERISK-26562</a>] - <td><td>app_controlplayback: Transmit Silence on ControlPlayback pause<br>(Reported by Mikheili Dautashvili)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24517'>ASTERISK-24517</a>] - <td><td>TLS support for Solaris, Ming and non-glibc Linux systems<br>(Reported by Timo Teräs)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26540'>ASTERISK-26540</a>] - <td><td>cdr_radius: use radcli instead of freeradius-client<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26558'>ASTERISK-26558</a>] - <td><td>app_queue: add variable to know if the call is not answered after a queue<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26176'>ASTERISK-26176</a>] - <td><td>chan_sip: Add AccountCode to AMI PeerEntry<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26217'>ASTERISK-26217</a>] - <td><td>[patch] Codec 2 Mode 2400<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26538'>ASTERISK-26538</a>] - <td><td>codec_opus: Add sample to configs/samples/codecs.conf.sample<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26488'>ASTERISK-26488</a>] - <td><td>ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26418'>ASTERISK-26418</a>] - <td><td>res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP<br>(Reported by Michael Walton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26422'>ASTERISK-26422</a>] - <td><td>[patch] Force calendars to do new fetch after module reload<br>(Reported by Ludovic Gasc (Eyepea))</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26398'>ASTERISK-26398</a>] - <td><td>core: Remove ABI differences of LOW_MEMORY<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26409'>ASTERISK-26409</a>] - <td><td>codec_opus: Update Asterisk to support the translation codec.<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26289'>ASTERISK-26289</a>] - <td><td>Announcer channels in ConfBridges cause inefficiencies<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26321'>ASTERISK-26321</a>] - <td><td>ARI : Add reason answered_elsewhere to channel hangup<br>(Reported by Jean Aunis - Prescom)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25980'>ASTERISK-25980</a>] - <td><td>[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26218'>ASTERISK-26218</a>] - <td><td>[patch] iLBC 20<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26190'>ASTERISK-26190</a>] - <td><td>[patch] SRTP: Enable AES-256 and AES-GCM.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26220'>ASTERISK-26220</a>] - <td><td>Add support for noreturn function attributes.<br>(Reported by Corey Farrell)</li></td></tr>
</table>
<p>
<b>Bugs fixed in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27152'>ASTERISK-27152</a>] - <td><td>Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27103'>ASTERISK-27103</a>] - <td><td>core: ast_safe_system command injection possible.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27013'>ASTERISK-27013</a>] - <td><td>res_rtp_asterisk: Media can be hijacked even with strict RTP enabled<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27231'>ASTERISK-27231</a>] - <td><td>res_rtp_asterisk: Allow remote SSRC to change due to renegotiation<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27222'>ASTERISK-27222</a>] - <td><td>core: Don't queue up multiple video update frames.<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27212'>ASTERISK-27212</a>] - <td><td>bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27202'>ASTERISK-27202</a>] - <td><td>If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed<br>(Reported by Seán C. McCord)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27200'>ASTERISK-27200</a>] - <td><td>manager: hook event is not being raised<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27179'>ASTERISK-27179</a>] - <td><td>res_pjsip_session: Handling of 'msid' is incorrect<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27182'>ASTERISK-27182</a>] - <td><td>bridge: Crash when mapping streams<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27189'>ASTERISK-27189</a>] - <td><td>Make --with-pjproject-bundled the default for Asterisk 15<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27180'>ASTERISK-27180</a>] - <td><td>channel: requester leaks joint_cap on success.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27171'>ASTERISK-27171</a>] - <td><td>Asterisk 15.0.0-Beta1 does not compile<br>(Reported by Ira Emus)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27119'>ASTERISK-27119</a>] - <td><td>res_pjsip: parse/add msid attribute when webrtc is enabled<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27143'>ASTERISK-27143</a>] - <td><td>bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27142'>ASTERISK-27142</a>] - <td><td>sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25810'>ASTERISK-25810</a>] - <td><td>say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.<br>(Reported by Nicolas Riendeau)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27136'>ASTERISK-27136</a>] - <td><td>bridge_softmix: Don't reorder SFU streams<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27134'>ASTERISK-27134</a>] - <td><td>bridge_softmix: Reuse any removed streams for video<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27133'>ASTERISK-27133</a>] - <td><td>res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27123'>ASTERISK-27123</a>] - <td><td>confbridge: Name recordings are left on filesystem<br>(Reported by Sergej Kasumovic)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27122'>ASTERISK-27122</a>] - <td><td>chan_iax2: On reload MWI taskprocessors keep adding up<br>(Reported by Sergej Kasumovic)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26807'>ASTERISK-26807</a>] - <td><td>sounds: New 3-D Binaural audio features require new sound prompts<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25816'>ASTERISK-25816</a>] - <td><td>French conf-adminmenu, conf-usermenu prompts differ in content from the English files<br>(Reported by Benoit Duverger)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26274'>ASTERISK-26274</a>] - <td><td>Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27118'>ASTERISK-27118</a>] - <td><td>res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27036'>ASTERISK-27036</a>] - <td><td>res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'<br>(Reported by Maxim Vasilev)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27023'>ASTERISK-27023</a>] - <td><td>res_rtp_asterisk: Deadlock when TURN session in use<br>(Reported by Jatin Jain)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27106'>ASTERISK-27106</a>] - <td><td>[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27093'>ASTERISK-27093</a>] - <td><td>ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting<br>(Reported by James Terhune)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27100'>ASTERISK-27100</a>] - <td><td>channel: ast_waitfordigit_full fails to clear flag in an error branch.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27090'>ASTERISK-27090</a>] - <td><td>PJSIP: Deadlock using TCP transport<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26997'>ASTERISK-26997</a>] - <td><td>Create an StreamEcho dialplan application<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27076'>ASTERISK-27076</a>] - <td><td>chan_pjsip: Add support for multiple streams<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27088'>ASTERISK-27088</a>] - <td><td>res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26978'>ASTERISK-26978</a>] - <td><td>rtp: Crash in ast_rtp_codecs_payload_code()<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25665'>ASTERISK-25665</a>] - <td><td>Duplicate logging in queue log for EXITEMPTY events<br>(Reported by Ove Aursand)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27065'>ASTERISK-27065</a>] - <td><td>call hangup after leaving app_queue<br>(Reported by Marek Cervenka)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24052'>ASTERISK-24052</a>] - <td><td>app_voicemail reloads result in leaked IMAP sockets.<br>(Reported by Louis Jocelyn Paquet)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27074'>ASTERISK-27074</a>] - <td><td>core_local: local channel data not being properly unref'ed and unlocked<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27075'>ASTERISK-27075</a>] - <td><td>bridge: stuck channel(s) after failed attended transfer<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27060'>ASTERISK-27060</a>] - <td><td>Comment typo format_g729.c<br>(Reported by Matthew Fredrickson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27041'>ASTERISK-27041</a>] - <td><td>Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br>(Reported by Frederic LE FOLL)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25370'>ASTERISK-25370</a>] - <td><td>res_corosync segfaults at startup with corosync version > 2.x<br>(Reported by mdu113)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27026'>ASTERISK-27026</a>] - <td><td>res_ari: Crash when no ari.conf configuration file exists<br>(Reported by Ronald Raikes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27016'>ASTERISK-27016</a>] - <td><td>Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.<br>(Reported by Chris Howard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27057'>ASTERISK-27057</a>] - <td><td>Seg Fault in ast_sorcery_object_get_id at sorcery.c<br>(Reported by Ryan Smith)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27024'>ASTERISK-27024</a>] - <td><td>nat/external_media settings ignored in 14.4.1<br>(Reported by Christopher van de Sande)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27022'>ASTERISK-27022</a>] - <td><td>res_rtp_asterisk: Incorrect SSRC change for RTCP component<br>(Reported by Michael Walton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26923'>ASTERISK-26923</a>] - <td><td>bridging: T.38 request is lost when channels are added to bridge<br>(Reported by Torrey Searle)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27053'>ASTERISK-27053</a>] - <td><td>res_pjsip_refer/session: Calls dropped during transfer<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27052'>ASTERISK-27052</a>] - <td><td>Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br>(Reported by alex)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27046'>ASTERISK-27046</a>] - <td><td>res_pjsip_transport_websocket: segfault in get_write_timeout<br>(Reported by Jørgen H)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27039'>ASTERISK-27039</a>] - <td><td>chan_pjsip: Device state is idle when channel from endpoint is in early media<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26996'>ASTERISK-26996</a>] - <td><td>chan_pjsip: Flipping between codecs<br>(Reported by Michael Maier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26281'>ASTERISK-26281</a>] - <td><td>chan_pjsip would send INVITE to 'Unreachable' endpoints<br>(Reported by Jacek Konieczny)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26973'>ASTERISK-26973</a>] - <td><td>bridge: Crash when freeing frame and snooping<br>(Reported by Michel R. Vaillancourt)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-19291'>ASTERISK-19291</a>] - <td><td>Background in realtime<br>(Reported by Andrew Nowrot)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27025'>ASTERISK-27025</a>] - <td><td>channel / meetme: Fix missing parentheses<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27021'>ASTERISK-27021</a>] - <td><td>GET /recordings/stored returns 500 Internal Server Error<br>(Reported by Tim Morgan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24858'>ASTERISK-24858</a>] - <td><td>[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br>(Reported by Frankie Chin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23951'>ASTERISK-23951</a>] - <td><td> Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25294'>ASTERISK-25294</a>] - <td><td>srtp's crypto_get_random deprecated<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23839'>ASTERISK-23839</a>] - <td><td>AGI - RECORD FILE - documentation doesn't describe BEEP argument<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22432'>ASTERISK-22432</a>] - <td><td>Async AGI crashes Asterisk when issuing "set variable" command without args<br>(Reported by Antoine Pitrou)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25662'>ASTERISK-25662</a>] - <td><td>Malformed AGI 520 Usage response<br>(Reported by Tony Mountifield)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27008'>ASTERISK-27008</a>] - <td><td>res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br>(Reported by John Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26399'>ASTERISK-26399</a>] - <td><td>app_queue: Agent not called when caller is parked<br>(Reported by wushumasters)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26400'>ASTERISK-26400</a>] - <td><td>app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26715'>ASTERISK-26715</a>] - <td><td>app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel<br>(Reported by David Brillert)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26975'>ASTERISK-26975</a>] - <td><td>app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call<br>(Reported by Lorne Gaetz)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27012'>ASTERISK-27012</a>] - <td><td>app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br>(Reported by Robert Mordec)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26979'>ASTERISK-26979</a>] - <td><td>res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br>(Reported by Javier Riveros )</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26982'>ASTERISK-26982</a>] - <td><td>chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br>(Reported by Stefan Engström)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26939'>ASTERISK-26939</a>] - <td><td>Out of bound memory access in PJSIP multipart parser crashes Asterisk<br>(Reported by Sandro Gauci)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26940'>ASTERISK-26940</a>] - <td><td>Asterisk Skinny memory exhaustion vulnerability leads to DoS<br>(Reported by Sandro Gauci)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26938'>ASTERISK-26938</a>] - <td><td>Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP<br>(Reported by Sandro Gauci)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26789'>ASTERISK-26789</a>] - <td><td>Audit manipulation of channel flags without locks<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26998'>ASTERISK-26998</a>] - <td><td>res_pjsip_session: INVITE retransmissions could still setup the same call again.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26143'>ASTERISK-26143</a>] - <td><td>res_rtp_asterisk: One way audio when transcoding<br>(Reported by Henning Holtschneider)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26333'>ASTERISK-26333</a>] - <td><td>Problems with Blind Transfer, PJSIP (Aastra 6869i)<br>(Reported by Matthias Binder)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26606'>ASTERISK-26606</a>] - <td><td>tcptls: Incorrect OpenSSL function call leads to misleading error report<br>(Reported by Bob Ham)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26983'>ASTERISK-26983</a>] - <td><td>Crash in Manager Reload when TLS Config Changes<br>(Reported by Joshua Elson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25032'>ASTERISK-25032</a>] - <td><td>[patch]cel_odbc sometimes inserts CEL with wrong eventtime<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26173'>ASTERISK-26173</a>] - <td><td>func_cdr: CDR function does not permit empty values to be assigned<br>(Reported by gkloepfer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25506'>ASTERISK-25506</a>] - <td><td>[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.<br>(Reported by Frederic LE FOLL)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24529'>ASTERISK-24529</a>] - <td><td>Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26966'>ASTERISK-26966</a>] - <td><td>bridge_simple: Add support for streams<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26974'>ASTERISK-26974</a>] - <td><td>res_pjsip: Deadlock in T.38 framehook<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26908'>ASTERISK-26908</a>] - <td><td>res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26959'>ASTERISK-26959</a>] - <td><td>dial: Allow topology of dialing channel to influence dialed channel<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25823'>ASTERISK-25823</a>] - <td><td>SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.<br>(Reported by Andreas Krüger)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26926'>ASTERISK-26926</a>] - <td><td>func_speex: Crash caused by frame with no datalen<br>(Reported by Richard Kenner)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26964'>ASTERISK-26964</a>] - <td><td>res_pjsip_session: Wrong From on reinvite when request and To URI differ<br>(Reported by Yasin CANER)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26930'>ASTERISK-26930</a>] - <td><td>pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux<br>(Reported by abelbeck)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26922'>ASTERISK-26922</a>] - <td><td>chan_sip: tcpbind uses wrong source address<br>(Reported by Ksenia)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26929'>ASTERISK-26929</a>] - <td><td>pjsip: Add database tables for RLS<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26949'>ASTERISK-26949</a>] - <td><td>sdp: Implement T.38<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26953'>ASTERISK-26953</a>] - <td><td>Asterisk crash if hep.conf have some missing parameters<br>(Reported by Joel Vandal)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26890'>ASTERISK-26890</a>] - <td><td>STUN server with non-default-route transport causes INVITE delay<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26951'>ASTERISK-26951</a>] - <td><td>chan_sip: ACK with SDP does not update a direct media bridge<br>(Reported by Jean Aunis - Prescom)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26692'>ASTERISK-26692</a>] - <td><td>res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26835'>ASTERISK-26835</a>] - <td><td>res_rtp_asterisk: Crash when freeing RTCP address string<br>(Reported by Niklas Larsson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26853'>ASTERISK-26853</a>] - <td><td>res_rtp_asterisk: Crash in pjnath when receiving packet<br>(Reported by Adagio)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26613'>ASTERISK-26613</a>] - <td><td>format_wav: wav16 format read file only by 320 - half of frame<br>(Reported by Vitaly K)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26169'>ASTERISK-26169</a>] - <td><td>format_ogg_vorbis: Memory leak using OGG in MixMonitor<br>(Reported by Ivan Myalkin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21856'>ASTERISK-21856</a>] - <td><td>STUN never works when asterisk started without internet access<br>(Reported by Jeremy Kister)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-20984'>ASTERISK-20984</a>] - <td><td>Audible clicks when playing sox encoded au file with STREAM FILE AGI command<br>(Reported by Roman S.)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26528'>ASTERISK-26528</a>] - <td><td>[UBSAN] strings.h:signed integer overflow in ast_str_case_hash<br>(Reported by Badalian Vyacheslav)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26851'>ASTERISK-26851</a>] - <td><td>res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport<br>(Reported by Richard Begg)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26903'>ASTERISK-26903</a>] - <td><td>Listening TCP/TLS sockets stop when temporarily out of open files<br>(Reported by Walter Doekes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26928'>ASTERISK-26928</a>] - <td><td>pjsip: Add database tables for PUBLISH support<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26927'>ASTERISK-26927</a>] - <td><td>pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26905'>ASTERISK-26905</a>] - <td><td>pjproject_bundled: Merge 3 upstream deadlock patches into bundled<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26920'>ASTERISK-26920</a>] - <td><td>app_queue: PAUSEALL/UNPAUSEALL does not log reason<br>(Reported by Troy Bowman)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26897'>ASTERISK-26897</a>] - <td><td>chan_sip: Security vulnerability with client code header<br>(Reported by Alex Villacís Lasso)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25974'>ASTERISK-25974</a>] - <td><td>Unused realtime MOH classes not purged on 'moh reload'<br>(Reported by Sébastien Couture)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26916'>ASTERISK-26916</a>] - <td><td>res_pjsip: Excessive refcount reached on transport ao2 object<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21721'>ASTERISK-21721</a>] - <td><td>SIP Failed to parse multiple Supported: headers<br>(Reported by Olle Johansson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26915'>ASTERISK-26915</a>] - <td><td>chan_sip: Session Timers required but refused wrongly.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26363'>ASTERISK-26363</a>] - <td><td>res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code<br>(Reported by Yaacov Akiba Slama)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26896'>ASTERISK-26896</a>] - <td><td>Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT<br>(Reported by twisted)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26705'>ASTERISK-26705</a>] - <td><td>libasteriskssl.so not found when asterisk is installed for the 1st time<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26900'>ASTERISK-26900</a>] - <td><td>sdp: Add support for connection address management and topology updating<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21009'>ASTERISK-21009</a>] - <td><td>xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client<br>(Reported by Marcello Ceschia)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25490'>ASTERISK-25490</a>] - <td><td>[patch]SDP crypto tag is validated incorrectly<br>(Reported by Joerg Sonnenberger)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26885'>ASTERISK-26885</a>] - <td><td>channel: Support dynamic number of file descriptors<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26086'>ASTERISK-26086</a>] - <td><td>res_musiconhold: format option is not documented adequately<br>(Reported by Jens Bürger)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23996'>ASTERISK-23996</a>] - <td><td>No core dumps because of res_musiconhold chdir.<br>(Reported by Walter Doekes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24712'>ASTERISK-24712</a>] - <td><td>xmpp: starttls problem causes connection spew<br>(Reported by Matthias Urlichs)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26814'>ASTERISK-26814</a>] - <td><td>pjproject_bundled build fails to download pjproject source when using cURL<br>(Reported by Gergely Dömsödi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23510'>ASTERISK-23510</a>] - <td><td>JABBER_STATUS fails with improper code 7 for unavailable clients<br>(Reported by Anthony Critelli)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21855'>ASTERISK-21855</a>] - <td><td>Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available<br>(Reported by Jeremy Kister)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25622'>ASTERISK-25622</a>] - <td><td>WARNING for "JABBER: socket read error" should be more specific<br>(Reported by Sean Darcy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26515'>ASTERISK-26515</a>] - <td><td>rtp_engine: Allocate RTP payloads on a per-session basis<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26818'>ASTERISK-26818</a>] - <td><td>cdr: Problem setting variables in h exten<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26850'>ASTERISK-26850</a>] - <td><td>res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets<br>(Reported by Max Norba)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26484'>ASTERISK-26484</a>] - <td><td>res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.<br>(Reported by Vinod Dharashive)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26776'>ASTERISK-26776</a>] - <td><td>res_pjsip_pubsub: Crash when generating xpidf content<br>(Reported by Andrew Green)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26880'>ASTERISK-26880</a>] - <td><td>Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled<br>(Reported by Kirsty Tyerman)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26875'>ASTERISK-26875</a>] - <td><td>app_mixmonitor: Recording out of sync when 183 but no RTP<br>(Reported by Aaron An)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26862'>ASTERISK-26862</a>] - <td><td>app_queue: Queue stops calling members with local interface after forwarding in previous call<br>(Reported by Robert Mordec)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26732'>ASTERISK-26732</a>] - <td><td>res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome<br>(Reported by Dan Jenkins)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26879'>ASTERISK-26879</a>] - <td><td>PJSIP external_media_address ignored if no local_net options are provided<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26867'>ASTERISK-26867</a>] - <td><td>autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).<br>(Reported by Krzysztof Trempala)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26869'>ASTERISK-26869</a>] - <td><td>res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension<br>(Reported by Torrey Searle)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26668'>ASTERISK-26668</a>] - <td><td>core: Malformed pattern matching extension (various factors) results in crash<br>(Reported by xrobau)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26865'>ASTERISK-26865</a>] - <td><td>chan_iax2: Reload of iax peer results in loss of host address/port<br>(Reported by Richard Begg)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26872'>ASTERISK-26872</a>] - <td><td>Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26717'>ASTERISK-26717</a>] - <td><td>Document the fact that Asterisk HEP support only works with the PJSIP channel driver<br>(Reported by Olivier Krief)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26643'>ASTERISK-26643</a>] - <td><td>Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk<br>(Reported by Roman Bedros)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25237'>ASTERISK-25237</a>] - <td><td> stasis_cache.c:845 caching_topic_exec: - misleading ERROR message<br>(Reported by Smirnov Aleksey)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26857'>ASTERISK-26857</a>] - <td><td>chan_pjsip: Dialplan function race condition<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26822'>ASTERISK-26822</a>] - <td><td>pjsip/cli_commands: pjsip show channelstats shows wrong codec<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26353'>ASTERISK-26353</a>] - <td><td>res_musiconhold: musiconhold seems to think that the general section is a class and issues warning<br>(Reported by Jonathan Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26685'>ASTERISK-26685</a>] - <td><td>res_pjsip: Crash when using IPv6 and Transport ws,wss<br>(Reported by Michael Balen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24562'>ASTERISK-24562</a>] - <td><td>app_voicemail: Cannot set fromstring on a per-mailbox basis<br>(Reported by Mark Scholten)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26842'>ASTERISK-26842</a>] - <td><td>Websocket becomes disconnected when trying to place call from browser<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26841'>ASTERISK-26841</a>] - <td><td>chan_sip: Call not cancelled after receiving a 422 response<br>(Reported by Jean Aunis - Prescom)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26839'>ASTERISK-26839</a>] - <td><td>core: Implement stream topology changing in channels<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26598'>ASTERISK-26598</a>] - <td><td>Saynumber is trying to get "and" from "digits/" subfolder<br>(Reported by Jonathan Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-17067'>ASTERISK-17067</a>] - <td><td>Long lines in call files cause spurious syntax error<br>(Reported by Dave Olszewski)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26796'>ASTERISK-26796</a>] - <td><td>res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'<br>(Reported by Jørgen H)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26816'>ASTERISK-26816</a>] - <td><td>Implement ast_read_stream in channels<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25628'>ASTERISK-25628</a>] - <td><td>res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging<br>(Reported by Dmitry Wagin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26774'>ASTERISK-26774</a>] - <td><td>core: Playback URL fails after some time<br>(Reported by Igor Gamayunov)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26825'>ASTERISK-26825</a>] - <td><td>pjsip.conf.sample: user_agent: still refers to branch 12<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26823'>ASTERISK-26823</a>] - <td><td>PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26623'>ASTERISK-26623</a>] - <td><td>res_pjsip: Crash when calling PJSIPShowEndpoint<br>(Reported by Jørgen H)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26808'>ASTERISK-26808</a>] - <td><td>res_pjsip_outbound_registration doesn't know about network change events<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26781'>ASTERISK-26781</a>] - <td><td>bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio<br>(Reported by Sean Bright)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26782'>ASTERISK-26782</a>] - <td><td>res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication<br>(Reported by Peter Sokolov)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26793'>ASTERISK-26793</a>] - <td><td>Implement ast_write_stream in channels<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26812'>ASTERISK-26812</a>] - <td><td>[patch] Fix download_externals To Allow The Use Of curl Or wget<br>(Reported by Michael L. Young)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-18271'>ASTERISK-18271</a>] - <td><td>Pattern matching with res_config_mysql extensions does not behave as expected<br>(Reported by Charlie Smurthwaite)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26811'>ASTERISK-26811</a>] - <td><td>stream: Add streams to "core show channel"<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-18731'>ASTERISK-18731</a>] - <td><td>[patch] DUNDi weight parameter not processed correctly<br>(Reported by Peter Racz)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26799'>ASTERISK-26799</a>] - <td><td>res_pjsip: Using an auth object for inbound and outbound authentication fails.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26669'>ASTERISK-26669</a>] - <td><td>PJSIP Segfault 13.13.1 (Bundled PJSIP)<br>(Reported by Nic Colledge)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26738'>ASTERISK-26738</a>] - <td><td>Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c<br>(Reported by Michael Maier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25893'>ASTERISK-25893</a>] - <td><td>Function vmauthenticate accesses uninitialized memory<br>(Reported by Filip Jenicek)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26580'>ASTERISK-26580</a>] - <td><td>[patch] Error during LDAP modify action when user unregisters<br>(Reported by Nicholas John Koch)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26802'>ASTERISK-26802</a>] - <td><td>[patch] Integrity Check Of PJSIP Download Fails<br>(Reported by Michael L. Young)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-15858'>ASTERISK-15858</a>] - <td><td>[patch] Fix query with double backslash in string literals and stop log warnings<br>(Reported by Humberto Figuera)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26057'>ASTERISK-26057</a>] - <td><td>res_config_sqlite3 uses incorrect query - unnecessary escape<br>(Reported by Stepan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23457'>ASTERISK-23457</a>] - <td><td>SQlite3: Realtime queue loading fails after PRAGMA query result<br>(Reported by Scott Griepentrog)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26794'>ASTERISK-26794</a>] - <td><td>http: Crash on Reload Only in ast_tcptls_server_start<br>(Reported by Joshua Elson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26714'>ASTERISK-26714</a>] - <td><td>Phone default have not ringing on ARM<br>(Reported by Igor Goncharovsky)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26696'>ASTERISK-26696</a>] - <td><td>pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh<br>(Reported by Zach R)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26756'>ASTERISK-26756</a>] - <td><td>res_pjsip_mwi: Asterisk does not terminate MWI subscription<br>(Reported by Carl Fortin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26790'>ASTERISK-26790</a>] - <td><td>Implement stream topology (non-change request) API usage in channels<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26723'>ASTERISK-26723</a>] - <td><td>VoiceMailPlayMsg not playing messages via realtime<br>(Reported by Ryan Rittgarn)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-18286'>ASTERISK-18286</a>] - <td><td>[patch] 'Silence' is truncated in Record()<br>(Reported by var)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26775'>ASTERISK-26775</a>] - <td><td>app_queue: reset abandoned in service level<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26786'>ASTERISK-26786</a>] - <td><td>Implement ast_stream_topology API<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26248'>ASTERISK-26248</a>] - <td><td>chan_pjsip: Error when calling PJSIP client with domain specified<br>(Reported by Norbert Varga)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26788'>ASTERISK-26788</a>] - <td><td>core: Protect flags during ast_waitfor<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26115'>ASTERISK-26115</a>] - <td><td>pbx: AMI Originate ignore "failed" extension on call failure<br>(Reported by Nasir Iqbal)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26773'>ASTERISK-26773</a>] - <td><td>stream: Add basic API<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26785'>ASTERISK-26785</a>] - <td><td>configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample<br>(Reported by Torrey Searle)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26772'>ASTERISK-26772</a>] - <td><td>Crash in srv.c on startup with pjsip<br>(Reported by nappsoft)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26770'>ASTERISK-26770</a>] - <td><td>res_stasis_device_state: Duplicate subscriptions when multiple received at same time<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26767'>ASTERISK-26767</a>] - <td><td>ARI channelvars cause memory leak<br>(Reported by Sébastien Duthil)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26716'>ASTERISK-26716</a>] - <td><td>ari: Channels with pre-dial handlers cannot be hung up via ARI<br>(Reported by Tom Pawelek)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26632'>ASTERISK-26632</a>] - <td><td>core: Possibility of a frame "imbalance" leading to stuck channels.<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25951'>ASTERISK-25951</a>] - <td><td>res_agi: run_agi eats frames it shouldn't<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26343'>ASTERISK-26343</a>] - <td><td>ASTERISK-25951 causes issues for callerid manipulation through agi<br>(Reported by Morten Tryfoss)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26704'>ASTERISK-26704</a>] - <td><td>res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.<br>(Reported by Anthony Messina)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26765'>ASTERISK-26765</a>] - <td><td>res_resolver_unbound: FRACK! Excessive ref count trap tripped.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-21094'>ASTERISK-21094</a>] - <td><td>MixMonitorMute mutes through stream if already slinear (e.g. Originate)<br>(Reported by David Woolley)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26679'>ASTERISK-26679</a>] - <td><td>Crash on invalid contact domain (pjsip aor)<br>(Reported by Dmitriy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26699'>ASTERISK-26699</a>] - <td><td>res_pjsip: Assertion when sending OPTIONS request to endpoint<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26754'>ASTERISK-26754</a>] - <td><td>build_tools: make_build_h does not handle \ in user name <br>(Reported by Kirill Katsnelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26755'>ASTERISK-26755</a>] - <td><td>app_queue: Random queues disappear on "core reload queue all"<br>(Reported by Kirill Katsnelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26735'>ASTERISK-26735</a>] - <td><td>res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect<br>(Reported by Michael Maier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26693'>ASTERISK-26693</a>] - <td><td>res_pjsip_endpoint_identifier_ip: Add support for SRV<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26743'>ASTERISK-26743</a>] - <td><td>PJPROJECT: Detecting compiled max log level does not work.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26731'>ASTERISK-26731</a>] - <td><td>res_sorcery_memory_cache: memory leak on every sorcery memory cache populate<br>(Reported by Ustinov Artem)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26739'>ASTERISK-26739</a>] - <td><td>voicemail API test: confuses expected and actual values<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26740'>ASTERISK-26740</a>] - <td><td>voicemail API test: uses varlibdir instead of datadir for a sound file<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26665'>ASTERISK-26665</a>] - <td><td>app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?<br>(Reported by Marek Cervenka)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26710'>ASTERISK-26710</a>] - <td><td>[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0<br>(Reported by Aaron An)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26672'>ASTERISK-26672</a>] - <td><td>Crash when setting remote address on RTP instance<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26670'>ASTERISK-26670</a>] - <td><td>[patch] Outgoing SIP-URI Dialing via PJSIP<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26691'>ASTERISK-26691</a>] - <td><td>Remember SDP negotiation on SIP_CODEC_INBOUND.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26673'>ASTERISK-26673</a>] - <td><td>chan_pjsip: Crash when using CHANNEL dialplan function around masquerade<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26684'>ASTERISK-26684</a>] - <td><td>res_pjsip: Various issues with compact SIP headers<br>(Reported by Joshua Elson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26683'>ASTERISK-26683</a>] - <td><td>res_calendar: Calendars duplicated after module reload<br>(Reported by Martin Tomec)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26655'>ASTERISK-26655</a>] - <td><td>[patch]pjsip: Transfers Broken with Compact Headers Enabled<br>(Reported by JoshE)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26621'>ASTERISK-26621</a>] - <td><td>app_queue: Queue application does not ring members with Local interface<br>(Reported by Jonas Kellens)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26586'>ASTERISK-26586</a>] - <td><td>chan_sip: Segfaults upon reload if client with MWI wasn't registered<br>(Reported by Michael Kuron)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25494'>ASTERISK-25494</a>] - <td><td>build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24499'>ASTERISK-24499</a>] - <td><td>Need more explicit debug when PJSIP dialstring is invalid<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25083'>ASTERISK-25083</a>] - <td><td>Message.c: Message channel becomes saturated with frames leading to spammy log messages<br>(Reported by Jonathan Rose)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26653'>ASTERISK-26653</a>] - <td><td>pjproject_bundled doesn't verify already downloaded tarballs<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26433'>ASTERISK-26433</a>] - <td><td>chan_sip: Allows To-tag checks to be bypassed, setting up new calls<br>(Reported by Walter Doekes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26579'>ASTERISK-26579</a>] - <td><td>codec_opus: Recursiveness when parsing fmtp line<br>(Reported by Jørgen H)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26644'>ASTERISK-26644</a>] - <td><td>PJSIPShowRegistrationsInbound just dumps all aors<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26647'>ASTERISK-26647</a>] - <td><td>Support older DNS style for OpenBSD<br>(Reported by snuffy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26490'>ASTERISK-26490</a>] - <td><td>res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"<br>(Reported by Juris Breicis)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26629'>ASTERISK-26629</a>] - <td><td>tests/manager: 4 test failures as a result of iostream change<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26109'>ASTERISK-26109</a>] - <td><td>Asterisk fails building with OpenSSL 1.1.0<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26617'>ASTERISK-26617</a>] - <td><td>res_rtp_asterisk: Can't bind on systems without IPv6<br>(Reported by Guido Falsi)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26603'>ASTERISK-26603</a>] - <td><td>[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24330'>ASTERISK-24330</a>] - <td><td>Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118<br>(Reported by Marek Cervenka)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26566'>ASTERISK-26566</a>] - <td><td>res_rtp_asterisk: RTT miscalculation in RTCP<br>(Reported by Hector Royo Concepcion)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26604'>ASTERISK-26604</a>] - <td><td>chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.<br>(Reported by Michael Kuron)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26608'>ASTERISK-26608</a>] - <td><td>Compile and link failures on OpenBSD<br>(Reported by snuffy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26520'>ASTERISK-26520</a>] - <td><td>codec_opus: Generated fmtp line has no content<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26605'>ASTERISK-26605</a>] - <td><td>codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26516'>ASTERISK-26516</a>] - <td><td>pjsip: Memory corruption with possible memory leak.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24515'>ASTERISK-24515</a>] - <td><td>Unconditional use of fopencookie() / funopen() is non-portable<br>(Reported by Timo Teräs)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26556'>ASTERISK-26556</a>] - <td><td>manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes<br>(Reported by Michelle Dupuis)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26592'>ASTERISK-26592</a>] - <td><td>Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26575'>ASTERISK-26575</a>] - <td><td>testsuite: Need to check PJSIP functionality when res_srtp is not loaded.<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26565'>ASTERISK-26565</a>] - <td><td>chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set<br>(Reported by Jason)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26573'>ASTERISK-26573</a>] - <td><td>Some typos in documentation of chan_sip.c<br>(Reported by C.J. Collier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26571'>ASTERISK-26571</a>] - <td><td>res_pjsip: Resolution incorrect when explicit IPv6 transport configured<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26468'>ASTERISK-26468</a>] - <td><td>ari: Bridge events stop working after this sequence of ARI calls<br>(Reported by Daniele Pallastrelli)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24400'>ASTERISK-24400</a>] - <td><td>ooh323 sends wrong hangup code<br>(Reported by Dmitry Melekhov)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26555'>ASTERISK-26555</a>] - <td><td>Multi-party Video: Fix some post Asterisk-11 regressions<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26412'>ASTERISK-26412</a>] - <td><td>build: Prepare for gcc 6.2<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26509'>ASTERISK-26509</a>] - <td><td>A few non-critical deprecation warnings when building on Ubuntu 16.10<br>(Reported by Jonathan Harris)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26523'>ASTERISK-26523</a>] - <td><td>chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression<br>(Reported by Michael Keuter)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26549'>ASTERISK-26549</a>] - <td><td>app_dial: When PickupChan() is used some channels may have incorrect device state<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24274'>ASTERISK-24274</a>] - <td><td>[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used<br>(Reported by Frankie Chin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26311'>ASTERISK-26311</a>] - <td><td>[patch] rtp_engine: Allow more than 32 dynamic payload types.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26546'>ASTERISK-26546</a>] - <td><td>mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26541'>ASTERISK-26541</a>] - <td><td>res_pjsip_sdp_rtp: Restrict number of formats to maximum<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26476'>ASTERISK-26476</a>] - <td><td>chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"<br>(Reported by Sergey Grachev)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25070'>ASTERISK-25070</a>] - <td><td>Fix FTBFS on Hurd<br>(Reported by Gabriele Giacone)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26537'>ASTERISK-26537</a>] - <td><td>AMI: NewConnectedLine event is not documented<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26526'>ASTERISK-26526</a>] - <td><td>[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy<br>(Reported by Badalian Vyacheslav)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26524'>ASTERISK-26524</a>] - <td><td>astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26344'>ASTERISK-26344</a>] - <td><td>Asterisk 13.11.0 + PJSIP crash<br>(Reported by Ian Gilmour)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26387'>ASTERISK-26387</a>] - <td><td>Asterisk segfaults shortly after starting even with no active calls. <br>(Reported by Harley Peters)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26506'>ASTERISK-26506</a>] - <td><td>[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c<br>(Reported by Matt Krokosz)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26513'>ASTERISK-26513</a>] - <td><td>tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26514'>ASTERISK-26514</a>] - <td><td>Super Awesome Company: Don't specify transport in pjsip.conf<br>(Reported by Rusty Newton)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26510'>ASTERISK-26510</a>] - <td><td>pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22480'>ASTERISK-22480</a>] - <td><td>Embedded pjproject: build.mak contains hardcoded full path to version.mak<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26480'>ASTERISK-26480</a>] - <td><td>[patch] CLI: core set debug: Auto-completes File not Module<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26307'>ASTERISK-26307</a>] - <td><td>res_pjsip_caller_id: Crash on outgoing change<br>(Reported by Bill Brigden)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26503'>ASTERISK-26503</a>] - <td><td>app_voicemail: Asterisk crashes when MailboxExists is used<br>(Reported by Doug Lytle)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26423'>ASTERISK-26423</a>] - <td><td>res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness<br>(Reported by Andreas Wetzel)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26309'>ASTERISK-26309</a>] - <td><td>[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26482'>ASTERISK-26482</a>] - <td><td>[patch] chan_pjsip: segfault on already disconnected session<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26455'>ASTERISK-26455</a>] - <td><td>cdr_radius / cel_radius: try fix memory leak<br>(Reported by Badalian Vyacheslav)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26421'>ASTERISK-26421</a>] - <td><td>Segmentation Fault with ARI originate into mixing bridge with 43 clients<br>(Reported by Andrew Nagy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26444'>ASTERISK-26444</a>] - <td><td>'features show' command in CLI does not return prompt.<br>(Reported by John Kiniston)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26356'>ASTERISK-26356</a>] - <td><td>menuselect: invalid test for GTK2<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26477'>ASTERISK-26477</a>] - <td><td>pjproject: SEGV during SSL operations<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26462'>ASTERISK-26462</a>] - <td><td>[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage<br>(Reported by Leandro Dardini)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26439'>ASTERISK-26439</a>] - <td><td>chan_rtp: Crash when originating<br>(Reported by Kayode)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-17470'>ASTERISK-17470</a>] - <td><td>[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.<br>(Reported by effie mouzeli)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26416'>ASTERISK-26416</a>] - <td><td>pjproject-bundled: configure fails to check for all required utilities<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26466'>ASTERISK-26466</a>] - <td><td>core: Be forgiving on external callerid that may be flawed so we don't drop events<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26362'>ASTERISK-26362</a>] - <td><td>res_config_mysql: Broken after 13.10<br>(Reported by Carlos Chavez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26446'>ASTERISK-26446</a>] - <td><td>app_dial: There's no way to override the hangupcause on unanswered channels<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26457'>ASTERISK-26457</a>] - <td><td>[patch] force_rport,auto_comedia: No NAT detection triggered.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26453'>ASTERISK-26453</a>] - <td><td>res_pjsip_config_wizard: Memory leak in module_unload<br>(Reported by Badalian Vyacheslav)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26410'>ASTERISK-26410</a>] - <td><td>core: Asterisk 14 doesn't show the header in the console or verbose when starting<br>(Reported by Dan Jenkins)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24311'>ASTERISK-24311</a>] - <td><td>Populating database via Alembic fails when using same database for multiple schema sets<br>(Reported by Dafi Ni)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26438'>ASTERISK-26438</a>] - <td><td>[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26330'>ASTERISK-26330</a>] - <td><td>app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26426'>ASTERISK-26426</a>] - <td><td>format_ogg_opus: remove from source<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-18232'>ASTERISK-18232</a>] - <td><td>Broken REGISTER sent to IPv4 server when bindaddr=[::]<br>(Reported by Jacek)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25468'>ASTERISK-25468</a>] - <td><td>Deadlock in chan_sip - core show locks shows do_monitor lock<br>(Reported by Barry Flanagan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26397'>ASTERISK-26397</a>] - <td><td>manager: PresenceState action crashes Asterisk 14<br>(Reported by Andrew Nagy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26389'>ASTERISK-26389</a>] - <td><td>res_odbc: Clean up pooling options<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26273'>ASTERISK-26273</a>] - <td><td>core: Won't compile when LOW_MEMORY is enabled<br>(Reported by Anthony Messina)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26391'>ASTERISK-26391</a>] - <td><td>Consoles do not display verbose logger messages even when requested.<br>(Reported by Marcelo Terres)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26352'>ASTERISK-26352</a>] - <td><td>Astcanary dies when doing "core restart"<br>(Reported by Walter Doekes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-19867'>ASTERISK-19867</a>] - <td><td>asterisk fails to lower its priority when astcanary dies<br>(Reported by Xavier Hienne)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26263'>ASTERISK-26263</a>] - <td><td>SQL error when using realtime and registering extension / inserting into ps_contacts<br>(Reported by Jeppe Ryskov Larsen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26365'>ASTERISK-26365</a>] - <td><td>rtp: Offer with multiple payloads for same codec is incorrectly handled<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26374'>ASTERISK-26374</a>] - <td><td>res_pjsip_multihomed: Contact port is rewritten for connectionful protocols<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26359'>ASTERISK-26359</a>] - <td><td>[patch] cdr_mysql: fails to use UTC if so instructed<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26367'>ASTERISK-26367</a>] - <td><td>rtp: Timestamps broken when video frame is across multiple RTP packets<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26375'>ASTERISK-26375</a>] - <td><td>res_pjsip_transport_management: Log message states seconds, but time value is milliseconds<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-19968'>ASTERISK-19968</a>] - <td><td>TCP Session-Timers not dropping call<br>(Reported by Aaron Hamstra)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26364'>ASTERISK-26364</a>] - <td><td>res_pjsip: Don't assume a request will have target addresses<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26360'>ASTERISK-26360</a>] - <td><td>app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26358'>ASTERISK-26358</a>] - <td><td>chan_sip: Contact is updated on re-200, but not on re-INVITE<br>(Reported by Walter Doekes)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26316'>ASTERISK-26316</a>] - <td><td>res_pjsip_callerid: Irregular URI causes unexpected callerid<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26349'>ASTERISK-26349</a>] - <td><td> 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed<br>(Reported by Dmitry Melekhov)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26317'>ASTERISK-26317</a>] - <td><td>res_pjsip_session: Add ability to use preferred codec only<br>(Reported by Aaron An)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26264'>ASTERISK-26264</a>] - <td><td>res_pjsip: Crash when applying ACL from non-existent endpoint<br>(Reported by nappsoft)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26272'>ASTERISK-26272</a>] - <td><td>chan_sip: File descriptors leak (UDP sockets)<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-20234'>ASTERISK-20234</a>] - <td><td>SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"<br>(Reported by tootai)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26341'>ASTERISK-26341</a>] - <td><td>ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26291'>ASTERISK-26291</a>] - <td><td>res_pjsip_session: segfault on already disconnected session<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23989'>ASTERISK-23989</a>] - <td><td>[patch]SDP offer/answer fails if crypto keys added to non-crypto offer<br>(Reported by Olle Johansson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25691'>ASTERISK-25691</a>] - <td><td>Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26331'>ASTERISK-26331</a>] - <td><td>Crash on “core show channeltype Surrogate” in ast_format_cap_get_names<br>(Reported by CGI.NET)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26085'>ASTERISK-26085</a>] - <td><td>app_mp3: results in timeout for streams<br>(Reported by Jens Bürger)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26269'>ASTERISK-26269</a>] - <td><td>res_pjsip: Wrong state for aors without registered contacts after startup<br>(Reported by nappsoft)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26226'>ASTERISK-26226</a>] - <td><td>pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26299'>ASTERISK-26299</a>] - <td><td>app_queue: Queue application sometimes stops calling members with Local interface<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26279'>ASTERISK-26279</a>] - <td><td>pjproject-bundled: Fails to compile on Debian 6<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26306'>ASTERISK-26306</a>] - <td><td>channel: Hang-up crashes, chan_pjsip not cleaning up properly<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26203'>ASTERISK-26203</a>] - <td><td>res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24822'>ASTERISK-24822</a>] - <td><td>Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code<br>(Reported by David Brillert)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22732'>ASTERISK-22732</a>] - <td><td>Deadlock potential in res_fax and CCSS with local channels.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26282'>ASTERISK-26282</a>] - <td><td>AEL: macro-call in Dial application, macro "lacks 's' extension"<br>(Reported by chris de rock)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22820'>ASTERISK-22820</a>] - <td><td>[patch] Plaintext auth is still supported in IAX2<br>(Reported by Eugene)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-22374'>ASTERISK-22374</a>] - <td><td>Finish mapping the sip.conf parameters to res_sip.conf parameters<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24425'>ASTERISK-24425</a>] - <td><td>[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)<br>(Reported by abelbeck)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26228'>ASTERISK-26228</a>] - <td><td>res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.<br>(Reported by Ali Ghavidel)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25472'>ASTERISK-25472</a>] - <td><td>Swagger scripts are not replacing format variable in file brief<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25984'>ASTERISK-25984</a>] - <td><td>res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it<br>(Reported by József Dudás)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26305'>ASTERISK-26305</a>] - <td><td>Asterisk 14: Two resolver unbound testsuite tests fail<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26288'>ASTERISK-26288</a>] - <td><td>followme: fails to reset config items to default values on reload<br>(Reported by Tzafrir Cohen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26303'>ASTERISK-26303</a>] - <td><td>[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25492'>ASTERISK-25492</a>] - <td><td>ARI: Path parameters are case sensitive<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26164'>ASTERISK-26164</a>] - <td><td>XMPP no longer triggers NOTIFY to device via chan_pjsip<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26233'>ASTERISK-26233</a>] - <td><td>pbx: Failure to remove inconsistent extension names<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26246'>ASTERISK-26246</a>] - <td><td>Security: Privilege escalation by AMI adding dialplan extensions.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26267'>ASTERISK-26267</a>] - <td><td>ast_register_atexit callbacks should be run on failed startup.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26241'>ASTERISK-26241</a>] - <td><td>res_pjsip: When using compact headers, rpid and pai are incorrectly generated<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25797'>ASTERISK-25797</a>] - <td><td>app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension<br>(Reported by Etienne Lessard)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26239'>ASTERISK-26239</a>] - <td><td>res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26238'>ASTERISK-26238</a>] - <td><td>res_pjsip: Empty global default_from_user causes crash<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26268'>ASTERISK-26268</a>] - <td><td>alembic: 'auth_username' not in PJSIP 'identify_by' enum<br>(Reported by Joshua Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26253'>ASTERISK-26253</a>] - <td><td>sdp_srtp: libsrtp now a required dependency, shouldn't be<br>(Reported by Ben Merrills)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26145'>ASTERISK-26145</a>] - <td><td>pjsip: Deadlock with suspend + masquerade + indicate<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26183'>ASTERISK-26183</a>] - <td><td>alembic: error when using sqlalchemy version 1.1.0b2<br>(Reported by Kevin Harwell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26283'>ASTERISK-26283</a>] - <td><td>res_resolver_unbound: fails configure on older Ubuntu and CentOS<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26280'>ASTERISK-26280</a>] - <td><td>DNS lookups can block channel media paths<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26278'>ASTERISK-26278</a>] - <td><td>asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25217'>ASTERISK-25217</a>] - <td><td>[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26265'>ASTERISK-26265</a>] - <td><td>Errors ignored from some parts of system initialization.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26206'>ASTERISK-26206</a>] - <td><td>[patch] res_pjsip: Use more compatible regex for get all<br>(Reported by Dmitry Wagin)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26256'>ASTERISK-26256</a>] - <td><td>[patch] SIP/SDP origin (o=) contains brackets with IP6<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25996'>ASTERISK-25996</a>] - <td><td>Remove "live_dangerously" requirement on DB(read)<br>(Reported by Andrew Nagy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26148'>ASTERISK-26148</a>] - <td><td>pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."<br>(Reported by Hans van Eijsden)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26237'>ASTERISK-26237</a>] - <td><td>Fax is detected on regular calls.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26227'>ASTERISK-26227</a>] - <td><td>sqlalchemy error due to long identifier name<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-14'>ASTERISK-14</a>] - <td><td>asterisk leaves zombie mpg123<br>(Reported by dcarr)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-23013'>ASTERISK-23013</a>] - <td><td>[patch] Deadlock between 'sip show channels' command and attended transfer handling<br>(Reported by Ben Smithurst)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26199'>ASTERISK-26199</a>] - <td><td>PJSIP: tx_data_destroy called twice<br>(Reported by Scott Griepentrog)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26166'>ASTERISK-26166</a>] - <td><td>res_pjsip_pubsub: Crash when decrementing reference count of message<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26174'>ASTERISK-26174</a>] - <td><td>res_pjsip: Crash when freeing cloned message in distributor<br>(Reported by Ross Beer)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26216'>ASTERISK-26216</a>] - <td><td>res_fax: Deadlock when detect fax while channel executing Playback<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26214'>ASTERISK-26214</a>] - <td><td>Allow arbitrary time for fax detection to end on a channel<br>(Reported by Richard Mudgett)</li></td></tr>
</table>
<p>
<b>New Features made in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27063'>ASTERISK-27063</a>] - <td><td>Add support for systemd socket activation<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27117'>ASTERISK-27117</a>] - <td><td>core: Add support for timelen parsing to ast_parse_arg and ACO.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27129'>ASTERISK-27129</a>] - <td><td>ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.<br>(Reported by Corey Farrell)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26995'>ASTERISK-26995</a>] - <td><td>Add QUEUE_FLOAT_PENALTY to app_queue<br>(Reported by Steve Davies)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26878'>ASTERISK-26878</a>] - <td><td>func_channel: Add ability to get the callid so dialplan has access to it.<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26863'>ASTERISK-26863</a>] - <td><td>res_pjsip: Add endpoint identification scheme based on a configured SIP header/value<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-17428'>ASTERISK-17428</a>] - <td><td>[patch] Allow "Comedian Mail" branding to be removed<br>(Reported by John Covert)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26584'>ASTERISK-26584</a>] - <td><td>[patch] RTCP feedback for codec modules<br>(Reported by Lorenzo Miniero)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-19862'>ASTERISK-19862</a>] - <td><td>app_queue: Update Data of Queues (use queues as outbound calls container)<br>(Reported by Sebastian Gutierrez)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26630'>ASTERISK-26630</a>] - <td><td>Make logging PJPROJECT messages a bit easier<br>(Reported by Richard Mudgett)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26587'>ASTERISK-26587</a>] - <td><td>app_originate: Add option to execute gosub prior to dial<br>(Reported by dkerr)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26595'>ASTERISK-26595</a>] - <td><td>ARI: Add the ability to control the source of video in a multi-party mixing bridge<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26492'>ASTERISK-26492</a>] - <td><td>ARI: Add ability to specify channel variables on websocket events<br>(Reported by Mark Michelson)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26470'>ASTERISK-26470</a>] - <td><td>ARI: Add an 'asterisk_id' field to outgoing events<br>(Reported by Matt Jordan)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26277'>ASTERISK-26277</a>] - <td><td>Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment<br>(Reported by Matt Jordan)</li></td></tr>
</table>
<p>
For a full list of changes in this release, please see the ChangeLog:<br>
<a href='http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0'>http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0</a>
<p>
<b>Thank you for your continued support of Asterisk!</b><br>
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