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Asterisk 15.7.0-rc1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 15.7.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 15.7.0-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:

Security bugs fixed in this release:
-----------------------------------

 

 Buffer overflow for DNS SRV/NAPTR records
(Reported by Jan Hoffmann)
 res_http_websocket: Crash when reading HTTP Upgrade requests
(Reported by Sean Bright)

 

Bugs fixed in this release:
-----------------------------------

 

 bridging: Asterisk crashes when receiving an empty realtime text frame
(Reported by Emmanuel BUU)
 app_queue: QueueMemberStatus Event flooding AMI
(Reported by Andrej)
 res_pjsip: improve realtime performance on CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
 app_queue: Queue member considered inuse after immediately hanging up during dialing.
(Reported by Cao Minh Hiep)
 stasis: Playing MOH to bridge with ARI does not work
(Reported by Cameron)
 res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
 chan_sip: SipNotify via AMI behaves differently to CLI
(Reported by Peter Katzmann)
 configure script does not enforce libunbound2 version
(Reported by Samuel Galarneau)
 testsuite: Sniffer assumes pjmedia will use ports below 10000
(Reported by Joshua C. Colp)
 rtp: Crash in off-nominal case where RTP instance can't be set up
(Reported by Lei Fu)
 PJSIP: Update bundled PJPROJECT to version 2.8
(Reported by Joshua C. Colp)
 res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
 chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs
(Reported by Will)
 res_pjproject build failure
(Reported by Jaco Kroon)
 chan_sip unstable with TLS after asterisk start or reloads
(Reported by David Hajek)
 [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
 channel.c: ARI ring only once
(Reported by Hajek Michal)
 Realtime queuemembers are not updated during retry phase
(Reported by lvl)
 alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean
(Reported by Joshua C. Colp)
 res_pjsip_transport_websocket: Properly set 'received' for IPv6
(Reported by Sean Bright)
 res_pjsip realtime: uri column in ps_contacts table can be too short
(Reported by Florian Floimair)

 

Improvements made in this release:
-----------------------------------

 

 Remove stale nonoptreq references
(Reported by Walter Doekes)

 

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0-rc1

Thank you for your continued support of Asterisk!