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The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.3.0.<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 18.3.0-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-29305'>ASTERISK-29305</a>] - <td><td>ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash<br>(Reported by Gregory Massel)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29260'>ASTERISK-29260</a>] - <td><td>sRTP Replay Protection ignored; even tears down long calls<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29227'>ASTERISK-29227</a>] - <td><td>res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash<br>(Reported by Ivan Poddubny)</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-29071'>ASTERISK-29071</a>] - <td><td>app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs<br>(Reported by Stefan Ruf)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29329'>ASTERISK-29329</a>] - <td><td>app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events<br>(Reported by N A)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-24434'>ASTERISK-24434</a>] - <td><td>Fix differing usage of assignment operators in modules.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-29306'>ASTERISK-29306</a>] - <td><td>strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition<br>(Reported by Vitezslav Novy)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29300'>ASTERISK-29300</a>] - <td><td>res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent<br>(Reported by Sebastian Damm)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29235'>ASTERISK-29235</a>] - <td><td>res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address<br>(Reported by Brian Paboojian)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29266'>ASTERISK-29266</a>] - <td><td>ICE Role conflict with an unauthorized session<br>(Reported by Salah Ahmed)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29105'>ASTERISK-29105</a>] - <td><td>chan_pjsip: 180 Ringing with SDP not changed into progress<br>(Reported by Sebastian Damm)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29297'>ASTERISK-29297</a>] - <td><td>say: Y2021 problem – Asterisk cannot say year 2021 in Dutch<br>(Reported by Jacek Konieczny)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29315'>ASTERISK-29315</a>] - <td><td>res_pjsip: re-registration gets stuck if setting initial auth credentials fails<br>(Reported by Nick French)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29312'>ASTERISK-29312</a>] - <td><td>res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters<br>(Reported by Alexei Gradinari)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-16799'>ASTERISK-16799</a>] - <td><td>Callee declined when 'beep' audio file does not exist<br>(Reported by IAMJames_)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29313'>ASTERISK-29313</a>] - <td><td>res_pjsip_refer: Segfault in progress notify<br>(Reported by George Joseph)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29293'>ASTERISK-29293</a>] - <td><td>res_config_pgsql: Limit realtime_pgsql() to return one (no more) record<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-29303'>ASTERISK-29303</a>] - <td><td>pjsip: Re-invite occurs when it shouldn't<br>(Reported by Benjamin Keith Ford)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29311'>ASTERISK-29311</a>] - <td><td>res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit<br>(Reported by Jaco Kroon)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28452'>ASTERISK-28452</a>] - <td><td>pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer<br>(Reported by Michael Maier)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29287'>ASTERISK-29287</a>] - <td><td>app.h: C++ compatibility broken<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-28369'>ASTERISK-28369</a>] - <td><td>app_queue: Member device state "invalid" when second call is ringing and hint is used<br>(Reported by Boolah )</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29203'>ASTERISK-29203</a>] - <td><td>res_pjsip_t38: Crash when changing state<br>(Reported by Gregory Massel)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29205'>ASTERISK-29205</a>] - <td><td>res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client<br>(Reported by Edvin Vidmar)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29196'>ASTERISK-29196</a>] - <td><td>res_pjsip: Segmentation fault<br>(Reported by Mauri de Souza Meneguzzo (3CPlus))</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29035'>ASTERISK-29035</a>] - <td><td>chan_local: Multistream support breaks T.38 faxing<br>(Reported by Matthias Hensler)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29280'>ASTERISK-29280</a>] - <td><td>chan_sip: Allow peers without audio (text+video).<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29265'>ASTERISK-29265</a>] - <td><td>chan_sip: Allow text+video media streams, again.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29261'>ASTERISK-29261</a>] - <td><td>res_pjsip: user=phone validation fail for isup numbers containing *#<br>(Reported by Mark Petersen)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29259'>ASTERISK-29259</a>] - <td><td>channel: Allow text+video media streams, again.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29258'>ASTERISK-29258</a>] - <td><td>chan_sip: Audio stream rejected, Other stream present: Invalid SDP.<br>(Reported by Alexander Traud)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29220'>ASTERISK-29220</a>] - <td><td>After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used<br>(Reported by Robert Cripps)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29248'>ASTERISK-29248</a>] - <td><td>res_pjsip_session: res sometimes uninitialized reported by compiler Clang.<br>(Reported by Alexander Traud)</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-29321'>ASTERISK-29321</a>] - <td><td>sorcery: Add support for more intelligent reloading.<br>(Reported by Joshua C. Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29325'>ASTERISK-29325</a>] - <td><td>res_pjsip_registrar: Include source IP address and port in log messages<br>(Reported by Joshua C. Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29326'>ASTERISK-29326</a>] - <td><td>asterisk: Update copyright/company<br>(Reported by Joshua C. Colp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29244'>ASTERISK-29244</a>] - <td><td>Add MixMonitorStart / Stop / Mute AMI events<br>(Reported by Sébastien Duthil)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29275'>ASTERISK-29275</a>] - <td><td>Support of MIME-type for wav16<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-29252'>ASTERISK-29252</a>] - <td><td>TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code<br>(Reported by Dan Cropp)</li></td></tr>
<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29262'>ASTERISK-29262</a>] - <td><td>Support of various URL-schemes by MoH<br>(Reported by Boris P. Korzun)</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-18.3.0-rc1'>https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0-rc1</a>
<p>
<b>Thank you for your continued support of Asterisk!</b><br>
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