[asterisk-users] Asterisk 17.3.0 Now Available
Asterisk Development Team
asteriskteam at digium.com
Thu Mar 12 11:46:41 CDT 2020
The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
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:
-----------------------------------
* ASTERISK-28766 - PJSIP blind transfer not completed after
using Proceeding()
(Reported by lvl)
* ASTERISK-28685 - check_expr2: linking (when hardening) and
cross-compiling troubles
(Reported by Sebastian Kemper)
* ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
seqno handling
(Reported by Joshua C. Colp)
* ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
the "variables" field
(Reported by Jean Aunis - Prescom)
* ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After
Hold
(Reported by Ross Beer)
* ASTERISK-28697 - res_pjsip: Named ACL does not update on
reload if changed
(Reported by Timothy Vanderaerden)
* ASTERISK-28746 - res_pjsip_outbound_registration keeps
retrying the first entry in a SRV record set
(Reported by
George Joseph)
* ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
complete before allowing sending
(Reported by Benjamin
Keith Ford)
* ASTERISK-28738 - Incorrect state machine used when
MOH_PASSTHRU is used
(Reported by Torrey Searle)
* ASTERISK-28742 - res_rtp_asterisk: static for audio due to
incomplete dtls/srtp setup
(Reported by Kevin Harwell)
* ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
to SLIN
(Reported by Ross Beer)
* ASTERISK-28730 - res_pjsip_session: Fix out of order session
refreshes
(Reported by Joshua C. Colp)
* ASTERISK-26955 - pjsip: SIP Packets with Via "received="
Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)
* ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
depleted, should return 503
(Reported by Walter Doekes)
* ASTERISK-28719 - Cannot remove defaultrule from queue using
realtime queues
(Reported by EDV O-TON)
* ASTERISK-28713 - res_stasis_playback: Error building JSON
(Reported by S��bastien Duthil)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes
Segfaults
(Reported by Ross Beer)
* ASTERISK-26082 - res_pjsip_messaging: MessageSend
Content-Type can't be changed
(Reported by Alex)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported
by Ross Beer)
* ASTERISK-28679 - stasis application is destroyed after its
creation
(Reported by Francois Blackburn)
* ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in
spite of the error when sending
(Reported by Dmitriy
Serov)
* ASTERISK-28686 - chan_sip strictrtp=yes fails when media
source is changed: no audio
(Reported by Walter Doekes)
* ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
Asterisk To Drop Calls
(Reported by Paul Brooks)
Improvements made in this release:
-----------------------------------
* ASTERISK-28750 - TLS/SSL Key too small error
(Reported
by Martin Zeh)
* ASTERISK-28733 - stream: Add support for adding/removing
streams during SFU/calls
(Reported by Joshua C. Colp)
* ASTERISK-24798 - Documentation - Clarify That Format Is Set
By File Name Extension In MixMonitor
(Reported by xrobau)
* ASTERISK-28726 - 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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