The Asterisk Development Team has announced the release of Asterisk 13.9.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.9.0 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:
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.
- [ASTERISK-25963] – func_odbc requires reconnect checks for stale connections
- [ASTERISK-25970] – Segfault in pjsip_url_compare
Improvement
- [ASTERISK-25444] – Music On Hold Warning misleading
- [ASTERISK-25865] – Message-Account Missing From PJSIP MWI
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0
Thank you for your continued support of Asterisk!