Asterisk 13.9.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.9.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.9.0-rc1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this release candidate:

Bug

  • [ASTERISK-24543] – Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
  • [ASTERISK-24596] – Unclear how to use Park application with res_parking ‘parkeddynamic’ enabled. Documentation?
  • [ASTERISK-24605] – res_parking option parkeddynamic does not work with the core Features ‘parkcall’ (DTMF initiated parking)
  • [ASTERISK-24649] – Pushing of channel into bridge fails; Stasis fails to get app name
  • [ASTERISK-24782] – StasisEnd event not present for channel that was swapped out for another after completing attended transfer
  • [ASTERISK-25123] – Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP
  • [ASTERISK-25407] – Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25510] – Log to syslog failing
  • [ASTERISK-25707] – Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions
  • [ASTERISK-25796] – res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES
  • [ASTERISK-25825] – Crashes during shutdown when running CLI commands
  • [ASTERISK-25826] – PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25854] – No audio after HOLD/RESUME – incorrect a=recvonly in SDP from Asterisk
  • [ASTERISK-25857] – func_aes: incorrect use of strlen() leads to data corruption
  • [ASTERISK-25867] – Video delay on app_echo
  • [ASTERISK-25873] – res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj
  • [ASTERISK-25874] – app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
  • [ASTERISK-25882] – ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)
  • [ASTERISK-25884] – unable to ./configure after ./bootstrap.sh
  • [ASTERISK-25885] – res_pjsip: Race condition between adding contact and automatic expiration
  • [ASTERISK-25888] – Frequent segfaults in function can_ring_entry() of app_queue.c
  • [ASTERISK-25890] – Asterisk 13.8.0 alembic database update fails
  • [ASTERISK-25894] – webrtc video broken due to missing marker bits in RTP streams
  • [ASTERISK-25910] – pjproject: Via headers are not parsed when “received” contains an IPv6 address
  • [ASTERISK-25912] – chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
  • [ASTERISK-25927] – Removed option “registertrying” is still documented in sip.conf.sample
  • [ASTERISK-25928] – res_pjsip: URI validation done outside of PJSIP thread
  • [ASTERISK-25929] – res_pjsip_registrar: AOR_CONTACT_ADDED events not raised
  • [ASTERISK-25934] – chan_sip should not require sipregs or updateable sippeers table unless rt
  • [ASTERISK-25938] – res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25942] – res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
  • [ASTERISK-25947] – Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object.

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0-rc1

Thank you for your continued support of Asterisk!

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