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The Asterisk Development Team would like to announce the first

release candidate of Asterisk 18.2.0.<br>

This release candidate is available for immediate download at <br>

<a href='https://downloads.asterisk.org/pub/telephony/asterisk'>https://downloads.asterisk.org/pub/telephony/asterisk</a>

<p>

The release of Asterisk 18.2.0-rc1 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 candidate:<br>

<p>

<b>Security 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-29219'>ASTERISK-29219</a>] - <td><td>res_pjsip_diversion: Crash if Tel URI contains History-Info<br>(Reported by Torrey Searle)</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-29229'>ASTERISK-29229</a>] - <td><td>Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription<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-29240'>ASTERISK-29240</a>] - <td><td>chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable<br>(Reported by Ivan Poddubny)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29238'>ASTERISK-29238</a>] - <td><td>chan_sip: SDP: Offers without any enabled stream are accepted.<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29237'>ASTERISK-29237</a>] - <td><td>chan_sip: SDP: m=video is parsed even when disabled.<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29222'>ASTERISK-29222</a>] - <td><td>chan_sip: Hold/Resume an sRTP call on a video enabled user-agent.<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-27902'>ASTERISK-27902</a>] - <td><td>chan_pjsip isn't updating hangupcause on 4XX responses<br>(Reported by George Joseph)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28016'>ASTERISK-28016</a>] - <td><td>PJSIP sends duplicate 183 Progress responses<br>(Reported by Alex Hermann)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28185'>ASTERISK-28185</a>] - <td><td>chan_pjsip: Subsequent same responses are not stopped<br>(Reported by Julien)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29230'>ASTERISK-29230</a>] - <td><td>pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send<br>(Reported by Michael Maier)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29231'>ASTERISK-29231</a>] - <td><td>pjsip: SIGSEGV in CLI if no trunk is registered<br>(Reported by Michael Maier)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29217'>ASTERISK-29217</a>] - <td><td>LOCK() can grant the same lock to multiple channels spuriously<br>(Reported by Jaco Kroon)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29201'>ASTERISK-29201</a>] - <td><td>Crash occurs when Transfer and execute Hangup before the Transfer result <br>(Reported by Dan Cropp)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28947'>ASTERISK-28947</a>] - <td><td>Segmentation fault in mixmonitor_ds_destroy<br>(Reported by Robert Sutton)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29168'>ASTERISK-29168</a>] - <td><td>Asterisk crashes during call transfer<br>(Reported by Dalius Mockevicius)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29210'>ASTERISK-29210</a>] - <td><td>res_pjsip: Crash when examining transport<br>(Reported by N GM )</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29191'>ASTERISK-29191</a>] - <td><td>tel: URI in Diversion header causes crash<br>(Reported by Mikhail Ivanov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28883'>ASTERISK-28883</a>] - <td><td>Spyee information ist missing in ChanSpyStop AMI Event<br>(Reported by Hendrik Wedhorn)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29188'>ASTERISK-29188</a>] - <td><td>null media causing the Asterisk crash<br>(Reported by sungtae kim)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29024'>ASTERISK-29024</a>] - <td><td>pjsip: Route Header in Cancel request incorrectly set<br>(Reported by Flole Systems)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29209'>ASTERISK-29209</a>] - <td><td>Debug messages printed by scope trace might be missing newlines<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29211'>ASTERISK-29211</a>] - <td><td>res_musiconhold: Segfault on realtime music on hold without entries<br>(Reported by Nathan Bruning)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29022'>ASTERISK-29022</a>] - <td><td>Crash when manipulating PJSIP invite dlg ref counts<br>(Reported by Sean Bright)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29173'>ASTERISK-29173</a>] - <td><td>Media cache URL requests allow infinite redirects<br>(Reported by Sean Bright)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29175'>ASTERISK-29175</a>] - <td><td>res_pjsip_stir_shaken: Fix module description<br>(Reported by Stanislav Abramenkov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29148'>ASTERISK-29148</a>] - <td><td>AST_MODULE_INFO no, MODULEINFO depend<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29165'>ASTERISK-29165</a>] - <td><td>res_pjsip: malformed header Accept-Encoding in OPTIONS response<br>(Reported by Alexander Greiner-Baer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28798'>ASTERISK-28798</a>] - <td><td>[patch] chan_sip: TCP/TLS client without server.<br>(Reported by Alexander Traud)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29161'>ASTERISK-29161</a>] - <td><td>Incorrect setup of recall channels<br>(Reported by Boris P. Korzun)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29155'>ASTERISK-29155</a>] - <td><td>app_queue: Deadlock between queues container and individual queues<br>(Reported by George Joseph)</li></td></tr>

</table>

<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-28549'>ASTERISK-28549</a>] - <td><td>Two repeated 183<br>(Reported by Gant Liu)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29216'>ASTERISK-29216</a>] - <td><td>contrib: systemd asterisk service for centos8 or other newer linux versions<br>(Reported by Mark Petersen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29143'>ASTERISK-29143</a>] - <td><td>res_http_media_cache: HTTP media cache stored hardcoded in /tmp<br>(Reported by laszlovl)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29118'>ASTERISK-29118</a>] - <td><td>VoiceMail() should have an option to play greetings as Early Media<br>(Reported by Juan Carlos Castro y Castro)</li></td></tr>

</table>

<p>

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

<a href='https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0-rc1'>https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0-rc1</a>

<p>

<b>Thank you for your continued support of Asterisk!</b><br>

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