<div dir="ltr"><span style="font-size:12.8px">The Asterisk Development Team would like to announce the release of Asterisk 14.4.0.</span><br style="font-size:12.8px"><span style="font-size:12.8px">This release is available for immediate download at</span><br style="font-size:12.8px"><a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank" style="font-size:12.8px">http://downloads.asterisk.org/<wbr>pub/telephony/asterisk</a><span style="font-size:12.8px"></span><p style="font-size:12.8px">The release of Asterisk 14.4.0 resolves several issues reported by the<br>community and would have not been possible without your participation.<br></p><p style="font-size:12.8px"><b>Thank you!</b><br></p><p style="font-size:12.8px">The following issues are resolved in this release:<br></p><p style="font-size:12.8px"><b>New Features made in this release:</b><br>------------------------------<wbr>-----<br></p><table border="0" style="font-size:12.8px"><tbody><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26878" target="_blank">ASTERISK-26878</a>] -</li></td><td></td><td>func_channel: Add ability to get the callid so dialplan has access to it.<br>(Reported by Richard Mudgett)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26863" target="_blank">ASTERISK-26863</a>] -</li></td><td></td><td>res_pjsip: Add endpoint identification scheme based on a configured SIP header/value<br>(Reported by Matt Jordan)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17428" target="_blank">ASTERISK-17428</a>] -</li></td><td></td><td>[patch] Allow "Comedian Mail" branding to be removed<br>(Reported by John Covert)</td></tr></tbody></table><p style="font-size:12.8px"><b>Bugs fixed in this release:</b><br>------------------------------<wbr>-----<br></p><table border="0" style="font-size:12.8px"><tbody><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26851" target="_blank">ASTERISK-26851</a>] -</li></td><td></td><td>res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport<br>(Reported by Richard Begg)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26897" target="_blank">ASTERISK-26897</a>] -</li></td><td></td><td>chan_sip: Security vulnerability with client code header<br>(Reported by Alex Villacís Lasso)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26916" target="_blank">ASTERISK-26916</a>] -</li></td><td></td><td>res_pjsip: Excessive refcount reached on transport ao2 object<br>(Reported by Ross Beer)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26705" target="_blank">ASTERISK-26705</a>] -</li></td><td></td><td>libasteriskssl.so not found when asterisk is installed for the 1st time<br>(Reported by George Joseph)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26850" target="_blank">ASTERISK-26850</a>] -</li></td><td></td><td>res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets<br>(Reported by Max Norba)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26484" target="_blank">ASTERISK-26484</a>] -</li></td><td></td><td>res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.<br>(Reported by Vinod Dharashive)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26776" target="_blank">ASTERISK-26776</a>] -</li></td><td></td><td>res_pjsip_pubsub: Crash when generating xpidf content<br>(Reported by Andrew Green)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26880" target="_blank">ASTERISK-26880</a>] -</li></td><td></td><td>Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled<br>(Reported by Kirsty Tyerman)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26862" target="_blank">ASTERISK-26862</a>] -</li></td><td></td><td>app_queue: Queue stops calling members with local interface after forwarding in previous call<br>(Reported by Robert Mordec)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26732" target="_blank">ASTERISK-26732</a>] -</li></td><td></td><td>res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome<br>(Reported by Dan Jenkins)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26879" target="_blank">ASTERISK-26879</a>] -</li></td><td></td><td>PJSIP external_media_address ignored if no local_net options are provided<br>(Reported by Matt Jordan)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26867" target="_blank">ASTERISK-26867</a>] -</li></td><td></td><td>autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).<br>(Reported by Krzysztof Trempala)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26869" target="_blank">ASTERISK-26869</a>] -</li></td><td></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)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26668" target="_blank">ASTERISK-26668</a>] -</li></td><td></td><td>core: Malformed pattern matching extension (various factors) results in crash<br>(Reported by xrobau)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26865" target="_blank">ASTERISK-26865</a>] -</li></td><td></td><td>chan_iax2: Reload of iax peer results in loss of host address/port<br>(Reported by Richard Begg)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26872" target="_blank">ASTERISK-26872</a>] -</li></td><td></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)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26717" target="_blank">ASTERISK-26717</a>] -</li></td><td></td><td>Document the fact that Asterisk HEP support only works with the PJSIP channel driver<br>(Reported by Olivier Krief)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26643" target="_blank">ASTERISK-26643</a>] -</li></td><td></td><td>Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk<br>(Reported by Roman Bedros)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25237" target="_blank">ASTERISK-25237</a>] -</li></td><td></td><td>stasis_cache.c:845 caching_topic_exec: - misleading ERROR message<br>(Reported by Smirnov Aleksey)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26857" target="_blank">ASTERISK-26857</a>] -</li></td><td></td><td>chan_pjsip: Dialplan function race condition<br>(Reported by Joshua Colp)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26841" target="_blank">ASTERISK-26841</a>] -</li></td><td></td><td>chan_sip: Call not cancelled after receiving a 422 response<br>(Reported by Jean Aunis - Prescom)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26822" target="_blank">ASTERISK-26822</a>] -</li></td><td></td><td>pjsip/cli_commands: pjsip show channelstats shows wrong codec<br>(Reported by Kevin Harwell)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26353" target="_blank">ASTERISK-26353</a>] -</li></td><td></td><td>res_musiconhold: musiconhold seems to think that the general section is a class and issues warning<br>(Reported by Jonathan Harris)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26685" target="_blank">ASTERISK-26685</a>] -</li></td><td></td><td>res_pjsip: Crash when using IPv6 and Transport ws,wss<br>(Reported by Michael Balen)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24562" target="_blank">ASTERISK-24562</a>] -</li></td><td></td><td>app_voicemail: Cannot set fromstring on a per-mailbox basis<br>(Reported by Mark Scholten)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26598" target="_blank">ASTERISK-26598</a>] -</li></td><td></td><td>Saynumber is trying to get "and" from "digits/" subfolder<br>(Reported by Jonathan Harris)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17067" target="_blank">ASTERISK-17067</a>] -</li></td><td></td><td>Long lines in call files cause spurious syntax error<br>(Reported by Dave Olszewski)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26796" target="_blank">ASTERISK-26796</a>] -</li></td><td></td><td>res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'<br>(Reported by Jørgen H)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25628" target="_blank">ASTERISK-25628</a>] -</li></td><td></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)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26774" target="_blank">ASTERISK-26774</a>] -</li></td><td></td><td>core: Playback URL fails after some time<br>(Reported by Igor Gamayunov)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26825" target="_blank">ASTERISK-26825</a>] -</li></td><td></td><td>pjsip.conf.sample: user_agent: still refers to branch 12<br>(Reported by Tzafrir Cohen)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26823" target="_blank">ASTERISK-26823</a>] -</li></td><td></td><td>PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist<br>(Reported by Mark Michelson)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26623" target="_blank">ASTERISK-26623</a>] -</li></td><td></td><td>res_pjsip: Crash when calling PJSIPShowEndpoint<br>(Reported by Jørgen H)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26808" target="_blank">ASTERISK-26808</a>] -</li></td><td></td><td>res_pjsip_outbound_registratio<wbr>n doesn't know about network change events<br>(Reported by George Joseph)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26781" target="_blank">ASTERISK-26781</a>] -</li></td><td></td><td>bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio<br>(Reported by Sean Bright)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26782" target="_blank">ASTERISK-26782</a>] -</li></td><td></td><td>res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication<br>(Reported by Peter Sokolov)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26812" target="_blank">ASTERISK-26812</a>] -</li></td><td></td><td>[patch] Fix download_externals To Allow The Use Of curl Or wget<br>(Reported by Michael L. Young)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18271" target="_blank">ASTERISK-18271</a>] -</li></td><td></td><td>Pattern matching with res_config_mysql extensions does not behave as expected<br>(Reported by Charlie Smurthwaite)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26669" target="_blank">ASTERISK-26669</a>] -</li></td><td></td><td>PJSIP Segfault 13.13.1 (Bundled PJSIP)<br>(Reported by Nic Colledge)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18731" target="_blank">ASTERISK-18731</a>] -</li></td><td></td><td>[patch] DUNDi weight parameter not processed correctly<br>(Reported by Peter Racz)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26799" target="_blank">ASTERISK-26799</a>] -</li></td><td></td><td>res_pjsip: Using an auth object for inbound and outbound authentication fails.<br>(Reported by Richard Mudgett)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26738" target="_blank">ASTERISK-26738</a>] -</li></td><td></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)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-25893" target="_blank">ASTERISK-25893</a>] -</li></td><td></td><td>Function vmauthenticate accesses uninitialized memory<br>(Reported by Filip Jenicek)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26580" target="_blank">ASTERISK-26580</a>] -</li></td><td></td><td>[patch] Error during LDAP modify action when user unregisters<br>(Reported by Nicholas John Koch)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26802" target="_blank">ASTERISK-26802</a>] -</li></td><td></td><td>[patch] Integrity Check Of PJSIP Download Fails<br>(Reported by Michael L. Young)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15858" target="_blank">ASTERISK-15858</a>] -</li></td><td></td><td>[patch] Fix query with double backslash in string literals and stop log warnings<br>(Reported by Humberto Figuera)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26057" target="_blank">ASTERISK-26057</a>] -</li></td><td></td><td>res_config_sqlite3 uses incorrect query - unnecessary escape<br>(Reported by Stepan)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23457" target="_blank">ASTERISK-23457</a>] -</li></td><td></td><td>SQlite3: Realtime queue loading fails after PRAGMA query result<br>(Reported by Scott Griepentrog)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26794" target="_blank">ASTERISK-26794</a>] -</li></td><td></td><td>http: Crash on Reload Only in ast_tcptls_server_start<br>(Reported by Joshua Elson)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26714" target="_blank">ASTERISK-26714</a>] -</li></td><td></td><td>Phone default have not ringing on ARM<br>(Reported by Igor Goncharovsky)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26696" target="_blank">ASTERISK-26696</a>] -</li></td><td></td><td>pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh<br>(Reported by Zach R)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26756" target="_blank">ASTERISK-26756</a>] -</li></td><td></td><td>res_pjsip_mwi: Asterisk does not terminate MWI subscription<br>(Reported by Carl Fortin)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26109" target="_blank">ASTERISK-26109</a>] -</li></td><td></td><td>Asterisk fails building with OpenSSL 1.1.0<br>(Reported by Tzafrir Cohen)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26723" target="_blank">ASTERISK-26723</a>] -</li></td><td></td><td>VoiceMailPlayMsg not playing messages via realtime<br>(Reported by Ryan Rittgarn)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18286" target="_blank">ASTERISK-18286</a>] -</li></td><td></td><td>[patch] 'Silence' is truncated in Record()<br>(Reported by var)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26248" target="_blank">ASTERISK-26248</a>] -</li></td><td></td><td>chan_pjsip: Error when calling PJSIP client with domain specified<br>(Reported by Norbert Varga)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26788" target="_blank">ASTERISK-26788</a>] -</li></td><td></td><td>core: Protect flags during ast_waitfor<br>(Reported by Joshua Colp)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26115" target="_blank">ASTERISK-26115</a>] -</li></td><td></td><td>pbx: AMI Originate ignore "failed" extension on call failure<br>(Reported by Nasir Iqbal)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26785" target="_blank">ASTERISK-26785</a>] -</li></td><td></td><td>configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample<br>(Reported by Torrey Searle)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26772" target="_blank">ASTERISK-26772</a>] -</li></td><td></td><td>Crash in srv.c on startup with pjsip<br>(Reported by nappsoft)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26770" target="_blank">ASTERISK-26770</a>] -</li></td><td></td><td>res_stasis_device_state: Duplicate subscriptions when multiple received at same time<br>(Reported by Joshua Colp)</td></tr></tbody></table><p style="font-size:12.8px"><b>Improvements made in this release:</b><br>------------------------------<wbr>-----<br></p><table border="0" style="font-size:12.8px"><tbody><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26864" target="_blank">ASTERISK-26864</a>] -</li></td><td></td><td>res_pjsip_session: Add support for overlap dialling<br>(Reported by Richard Begg)</td></tr><tr><td valign="top" nowrap><li style="margin-left:15px">[<a href="https://issues.asterisk.org/jira/browse/ASTERISK-26846" target="_blank">ASTERISK-26846</a>] -</li></td><td></td><td>chan_sip: Add rtcp-mux support<br>(Reported by Sean Bright)</td></tr></tbody></table><p style="font-size:12.8px">For a full list of changes in this release, please see the ChangeLog:<br><a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.4.0" target="_blank">http://downloads.asterisk.org/<wbr>pub/telephony/asterisk/ChangeL<wbr>og-14.4.0</a></p><p style="font-size:12.8px"><b>Thank you for your continued support of Asterisk!</b></p><br clear="all"><div><br></div>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><pre><span>Digium's Asterisk Development Team</span>

<span>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></span></pre></div></div>

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