[asterisk-dev] Asterisk 13.24.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.24.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.24.0-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-28013 - res_http_websocket: Crash when reading HTTP
      Upgrade requests
      (Reported by Sean Bright)

New Features made in this release:
-----------------------------------
 * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
      in Contact header in chan_pjsip
      (Reported by Torrey
      Searle)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28125 - app_queue: Revert broken queue channel
      reference patch
      (Reported by lvl)
 * ASTERISK-28151 - app_voicemail: MWI fails with
      mailboxes=##@device instead of mailboxes=##@default
     
      (Reported by Ronald Raikes)
 * ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
      modules unload
      (Reported by sungtae kim)
 * ASTERISK-28159 - SIGABRT caused by stack corruption in
      hashkeys_read when no matching keys present
      (Reported by
      Michael Walton)
 * ASTERISK-28140 - repeated segmentation faults 
     
      (Reported by Eyal Hasson)
 * ASTERISK-28103 - stasis: Filter messages at publishing to
      reduce work done
      (Reported by Joshua C. Colp)
 * ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
      Re-Invite omits routset
      (Reported by Torrey Searle)
 * ASTERISK-28158 - Some conditions prevent running of el_end,
      break the terminal.
      (Reported by Corey Farrell)
 * ASTERISK-28162 - [patch] need to reset DTMF last sequence
      number and timestamp on voice packet with marker bit
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28110 - rtp: Incorrect Packetization
      (Reported
      by Robert Cripps)
 * ASTERISK-28146 - pbx_config: Only the first [globals] section
      is processed.
      (Reported by Corey Farrell)
 * ASTERISK-28150 - Formatting error in documentation
     
      (Reported by Scott Griepentrog)
 * ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
      report AST_CEL_PICKUP in handle_invite_replaces
      (Reported
      by Luit van Drongelen)
 * ASTERISK-28137 - res_pjsip_notify: improve realtime
      performance on CLI completion on the endpoint
      (Reported by
      Alexei Gradinari)
 * ASTERISK-27980 - Caller ID cannot be changed on Attended
      Transfer before dialing out
      (Reported by Alexei Gradinari)
 * ASTERISK-28089 - function ast_sendtext() create RTP realtime
      packets with a trailing null byte in the payload
      (Reported
      by Emmanuel BUU)
 * 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-26094 - stasis: Playing MOH to bridge with ARI does
      not work
      (Reported by Cameron)
 * ASTERISK-27920 - app_queue: Queue member considered inuse
      after immediately hanging up during dialing.
      (Reported by
      Cao Minh Hiep)
 * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
      ports below 10000
      (Reported by Joshua C. Colp)
 * ASTERISK-28065 - res_odbc: missing SQL error diagnostic
     
      (Reported by Alexei Gradinari)
 * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
     
      (Reported by Sergej Kasumovic)
 * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
      2.8
      (Reported by Joshua C. Colp)
 * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
      differently to CLI
      (Reported by Peter Katzmann)
 * ASTERISK-28049 - res_pjproject build failure
      (Reported
      by Jaco Kroon)
 * 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-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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
      parse an URI and return a specified part of the URI
     
      (Reported by Alexei Gradinari)
 * ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
      a pipe
      (Reported by Pascal Cadotte Michaud)
 * 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-13.24.0-rc1

Thank you for your continued support of Asterisk!
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