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The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.<br>

This release is available for immediate download at<br>

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

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The release of Asterisk 17.3.0 resolves several issues reported by the<br>

community and would have not been possible without your participation.<br>

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<b>Thank you!</b><br>

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The following issues are resolved in this release:<br>

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<b>Bugs fixed 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-28766'>ASTERISK-28766</a>] - <td><td>PJSIP blind transfer not completed after using Proceeding()<br>(Reported by lvl)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28685'>ASTERISK-28685</a>] - <td><td>check_expr2: linking (when hardening) and cross-compiling troubles<br>(Reported by Sebastian Kemper)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28764'>ASTERISK-28764</a>] - <td><td>res_rtp_asterisk: Improve NACK support and seqno handling<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-28755'>ASTERISK-28755</a>] - <td><td>SIP/Stasis: SIP headers not transmitted in the "variables" field<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-28754'>ASTERISK-28754</a>] - <td><td>ASTERISK-28738 Causes Audio Issue After Hold<br>(Reported by Ross Beer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28697'>ASTERISK-28697</a>] - <td><td>res_pjsip: Named ACL does not update on reload if changed<br>(Reported by Timothy Vanderaerden)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28746'>ASTERISK-28746</a>] - <td><td>res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set<br>(Reported by George Joseph)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28716'>ASTERISK-28716</a>] - <td><td>ICE: pjnath shouldn't wait for ICE to complete before allowing sending<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-28738'>ASTERISK-28738</a>] - <td><td>Incorrect state machine used when MOH_PASSTHRU is used<br>(Reported by Torrey Searle)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28742'>ASTERISK-28742</a>] - <td><td>res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup<br>(Reported by Kevin Harwell)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28735'>ASTERISK-28735</a>] - <td><td>Realtime MoH Unknown format '' -- defaulting to SLIN<br>(Reported by Ross Beer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28730'>ASTERISK-28730</a>] - <td><td>res_pjsip_session: Fix out of order session refreshes<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-26955'>ASTERISK-26955</a>] - <td><td>pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected<br>(Reported by Peter Sokolov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28718'>ASTERISK-28718</a>] - <td><td>chan_sip: Returns 403 if RTP ports are depleted, should return 503<br>(Reported by Walter Doekes)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28719'>ASTERISK-28719</a>] - <td><td>Cannot remove defaultrule from queue using realtime queues<br>(Reported by EDV O-TON)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28713'>ASTERISK-28713</a>] - <td><td>res_stasis_playback: Error building JSON<br>(Reported by Sébastien Duthil)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28714'>ASTERISK-28714</a>] - <td><td>REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults<br>(Reported by Ross Beer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-26082'>ASTERISK-26082</a>] - <td><td>res_pjsip_messaging: MessageSend Content-Type can't be changed<br>(Reported by Alex)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28423'>ASTERISK-28423</a>] - <td><td>ARI causes STASIS Deadlock<br>(Reported by Ross Beer)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28679'>ASTERISK-28679</a>] - <td><td>stasis application is destroyed after its creation<br>(Reported by Francois Blackburn)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-25421'>ASTERISK-25421</a>] - <td><td>PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending<br>(Reported by Dmitriy Serov)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28686'>ASTERISK-28686</a>] - <td><td>chan_sip strictrtp=yes fails when media source is changed: no audio<br>(Reported by Walter Doekes)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28139'>ASTERISK-28139</a>] - <td><td>RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls<br>(Reported by Paul Brooks)</li></td></tr>

</table>

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<b>Improvements 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-28750'>ASTERISK-28750</a>] - <td><td>TLS/SSL Key too small error<br>(Reported by Martin Zeh)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28733'>ASTERISK-28733</a>] - <td><td>stream: Add support for adding/removing streams during SFU/calls<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-24798'>ASTERISK-24798</a>] - <td><td>Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor<br>(Reported by xrobau)</li></td></tr>

<tr><td valign=top nowrap='nowrap'><li>[<a href='https://issues.asterisk.org/jira/browse/ASTERISK-28726'>ASTERISK-28726</a>] - <td><td>install_prereq script uses the interactive mode when installing aptitude<br>(Reported by Sylvain Afchain)</li></td></tr>

</table>

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

<a href='https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0'>https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0</a>

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

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