The release of Asterisk 17.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Bugs fixed in this release:
-----------------------------------
| PJSIP blind transfer not completed after using Proceeding() | (Reported by lvl) | |
| check_expr2: linking (when hardening) and cross-compiling troubles | (Reported by Sebastian Kemper) | |
| res_rtp_asterisk: Improve NACK support and seqno handling | (Reported by Joshua C. Colp) | |
| SIP/Stasis: SIP headers not transmitted in the "variables" field | (Reported by Jean Aunis - Prescom) | |
| ASTERISK-28738 Causes Audio Issue After Hold | (Reported by Ross Beer) | |
| res_pjsip: Named ACL does not update on reload if changed | (Reported by Timothy Vanderaerden) | |
| res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set | (Reported by George Joseph) | |
| ICE: pjnath shouldn't wait for ICE to complete before allowing sending | (Reported by Benjamin Keith Ford) | |
| Incorrect state machine used when MOH_PASSTHRU is used | (Reported by Torrey Searle) | |
| res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup | (Reported by Kevin Harwell) | |
| Realtime MoH Unknown format '' -- defaulting to SLIN | (Reported by Ross Beer) | |
| res_pjsip_session: Fix out of order session refreshes | (Reported by Joshua C. Colp) | |
| pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected | (Reported by Peter Sokolov) | |
| chan_sip: Returns 403 if RTP ports are depleted, should return 503 | (Reported by Walter Doekes) | |
| Cannot remove defaultrule from queue using realtime queues | (Reported by EDV O-TON) | |
| res_stasis_playback: Error building JSON | (Reported by Sébastien Duthil) | |
| REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults | (Reported by Ross Beer) | |
| res_pjsip_messaging: MessageSend Content-Type can't be changed | (Reported by Alex) | |
| ARI causes STASIS Deadlock | (Reported by Ross Beer) | |
| stasis application is destroyed after its creation | (Reported by Francois Blackburn) | |
| PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending | (Reported by Dmitriy Serov) | |
| chan_sip strictrtp=yes fails when media source is changed: no audio | (Reported by Walter Doekes) | |
| RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls | (Reported by Paul Brooks) |
Improvements made in this release:
-----------------------------------
| TLS/SSL Key too small error | (Reported by Martin Zeh) | |
| stream: Add support for adding/removing streams during SFU/calls | (Reported by Joshua C. Colp) | |
| Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor | (Reported by xrobau) | |
| install_prereq script uses the interactive mode when installing aptitude | (Reported by Sylvain Afchain) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
Thank you for your continued support of Asterisk!
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