[asterisk-dev] Asterisk 17.3.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Mar 5 12:09:46 CST 2020


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.3.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.3.0-rc1 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 candidate:

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 candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0-rc1

Thank you for your continued support of Asterisk!
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