[asterisk-dev] Asterisk 15.7.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 15.7.0.
This release candidate 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 candidate:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
  
      (Reported by Jan Hoffmann)
 * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
      Upgrade requests
      (Reported by Sean Bright)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28076 - bridging: Asterisk crashes when receiving an
      empty realtime text frame
      (Reported by Emmanuel BUU)
 * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
      AMI
      (Reported by Andrej)
 * ASTERISK-28077 - res_pjsip: improve realtime performance on
      CLI 'pjsip show contacts'
      (Reported by Alexei Gradinari)
 * ASTERISK-27920 - app_queue: Queue member considered inuse
      after immediately hanging up during dialing.
      (Reported by
      Cao Minh Hiep)
 * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
      not work
      (Reported by Cameron)
 * ASTERISK-28065 - res_odbc: missing SQL error diagnostic
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
      differently to CLI
      (Reported by Peter Katzmann)
 * ASTERISK-28045 - configure script does not enforce
      libunbound2 version
      (Reported by Samuel Galarneau)
 * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
      ports below 10000
      (Reported by Joshua C. Colp)
 * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
      instance can't be set up
      (Reported by Lei Fu)
 * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
      2.8
      (Reported by Joshua C. Colp)
 * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
     
      (Reported by Sergej Kasumovic)
 * ASTERISK-28047 - chan_pjsip: Declined video stream is added
      when no video codecs configured and session refresh with removed
      video stream occurs
      (Reported by Will)
 * ASTERISK-28049 - res_pjproject build failure
      (Reported
      by Jaco Kroon)
 * ASTERISK-28034 - chan_sip unstable with TLS after asterisk
      start or reloads
      (Reported by David Hajek)
 * ASTERISK-28029 - [patch] res_musiconhold : music on hold will
      not start if previous hold just reached end of file
     
      (Reported by Frederic LE FOLL)
 * ASTERISK-28005 - channel.c: ARI ring only once
     
      (Reported by Hajek Michal)
 * ASTERISK-28032 - Realtime queuemembers are not updated during
      retry phase
      (Reported by lvl)
 * ASTERISK-27988 - alembic: PJSIP
      "mwi_subscribe_replaces_unsolicited" field is integer not
      boolean
      (Reported by Joshua C. Colp)
 * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
      'received' for IPv6
      (Reported by Sean Bright)
 * ASTERISK-28022 - res_pjsip realtime: uri column in
      ps_contacts table can be too short
      (Reported by Florian
      Floimair)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28046 - Remove stale nonoptreq references
     
      (Reported by Walter Doekes)

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

Thank you for your continued support of Asterisk!
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