[asterisk-users] Asterisk 16.9.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Mar 12 11:13:03 CDT 2020

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
This release is available for immediate download at

The release of Asterisk 16.9.0 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
      (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
      (Reported by Joshua C. Colp)
 * 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
      (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
      by Ross Beer)
 * ASTERISK-28679 - stasis application is destroyed after its
      (Reported by Francois Blackburn)
      spite of the error when sending
      (Reported by Dmitriy
 * 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)
 * ASTERISK-26955 - pjsip: SIP Packets with Via "received="
      Containing IPv6 Address Delimited by "[]" Rejected
      (Reported by Peter Sokolov)

Improvements made in this release:
 * ASTERISK-28750 - TLS/SSL Key too small error
      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

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

Thank you for your continued support of Asterisk!
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