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The Asterisk Development Team would like to announce the first

release candidate of Asterisk 18.17.0.<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 18.17.0-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>New Features made in this release:</b><br>

-----------------------------------<br>

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<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29810'>ASTERISK-29810</a>] - <td><td>app_signal: Add channel signaling applications<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30262'>ASTERISK-30262</a>] - <td><td>res_pjsip_session: Allow a context to be specified for overlap dialing<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30319'>ASTERISK-30319</a>] - <td><td>Add BYE Reason support for SIP<br>(Reported by Igor Goncharovsky)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30180'>ASTERISK-30180</a>] - <td><td>app_broadcast: Add a channel audio multicasting application<br>(Reported by N A)</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-27830'>ASTERISK-27830</a>] - <td><td>Asterisk crashes on Invalid UTF-8 string<br>(Reported by AvayaXAsterisk)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30354'>ASTERISK-30354</a>] - <td><td>chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30162'>ASTERISK-30162</a>] - <td><td>when chan_iax is used to relay calls, no ringing indication is played<br>(Reported by Jaco Kroon)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30424'>ASTERISK-30424</a>] - <td><td>pjproject_bundled: cross-compilation broken when ssl autodetected<br>(Reported by Nick French)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30388'>ASTERISK-30388</a>] - <td><td>res_phoneprov: Stale SERVER variable when multi-homed<br>(Reported by cmaj)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30419'>ASTERISK-30419</a>] - <td><td>pjsip: Crash when sending NOTIFY in PJSIP 2.13<br>(Reported by Ross Beer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30417'>ASTERISK-30417</a>] - <td><td>Copy/Paste error in UnpauseQueueMember<br>(Reported by Sean Bright)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30406'>ASTERISK-30406</a>] - <td><td>pbx_ael: Global variables are not expanded.<br>(Reported by Sean Bright)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29604'>ASTERISK-29604</a>] - <td><td>ari: Segfault with lots of calls<br>(Reported by Danila Evgrafov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30391'>ASTERISK-30391</a>] - <td><td>res_rtp_asterisk: Issue with transcoding g722 after MES changes<br>(Reported by George Joseph)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30345'>ASTERISK-30345</a>] - <td><td>loader.c: Modules that decline to load cannot be reloaded<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30379'>ASTERISK-30379</a>] - <td><td>http: fix NULL pointer dereference while enable_status on TLS-only<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-30375'>ASTERISK-30375</a>] - <td><td>res_http_media_cache: Crash when URL has no path component.<br>(Reported by Sean Bright)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30351'>ASTERISK-30351</a>] - <td><td>manager: Originate variables are not added when setvar used in manager.conf<br>(Reported by Sebastian Gutierrez)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30369'>ASTERISK-30369</a>] - <td><td>res_pjsip: Websockets from same IP shut down when they shouldn't be<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-30367'>ASTERISK-30367</a>] - <td><td>pbx: Fix outdated channel snapshots with pbx_exec<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28767'>ASTERISK-28767</a>] - <td><td>chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late<br>(Reported by Oleg)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30350'>ASTERISK-30350</a>] - <td><td>res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold<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-30240'>ASTERISK-30240</a>] - <td><td>app voicemail odbc build error with gcc 11.1<br>(Reported by Michael Bradeen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30100'>ASTERISK-30100</a>] - <td><td>res_pjsip: Path is ignored on INVITE to endpoint<br>(Reported by Yury Kirsanov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30198'>ASTERISK-30198</a>] - <td><td>Error `Too many open files` occurs after about ~8000 calls when using mixmonitor<br>(Reported by Julien Alie)</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-30411'>ASTERISK-30411</a>] - <td><td>app_read: add option to include terminating digit on empty, terminated strings<br>(Reported by Michael Bradeen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30405'>ASTERISK-30405</a>] - <td><td>app_directory: Add 's' option to skip channel call<br>(Reported by Michael Bradeen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30422'>ASTERISK-30422</a>] - <td><td>app_senddtmf: add the option for senddtmf to answer<br>(Reported by Michael Bradeen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30325'>ASTERISK-30325</a>] - <td><td>Upgrade Asterisk to bundled pjproject 2.13<br>(Reported by Stanislav Abramenkov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30404'>ASTERISK-30404</a>] - <td><td>app_directory: Add reading directory configuration from custom file<br>(Reported by Michael Bradeen)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-29913'>ASTERISK-29913</a>] - <td><td>func_json: Adds multi-level and array parsing to JSON_DECODE<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30353'>ASTERISK-30353</a>] - <td><td>func_frame_trace: Print text for text frames<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30361'>ASTERISK-30361</a>] - <td><td>json.h: Add missing ast_json_object_real_get<br>(Reported by N A)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30280'>ASTERISK-30280</a>] - <td><td>Create capability to assign a Media Experience Score to RTP streams<br>(Reported by George Joseph)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-30332'>ASTERISK-30332</a>] - <td><td>func_callerid: Warn if invalid redirecting reason provided<br>(Reported by N A)</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-18.17.0-rc1'>https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0-rc1</a>

<p>

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

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