The Asterisk Development Team has announced the first release candidate of
Asterisk 12.6.0. This release candidate is available for immediate
download at

The release of Asterisk 12.6.0-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:

Bugs fixed in this release:
 * ASTERISK-24027 - MixMonitor AMI action called during AGI
      execution from bridge feature causes channel to leave AGI has
      hung up (Reported by Matt Jordan)
 * ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends on
      pjsip (Reported by Matt Jordan)
 * ASTERISK-24032 - Gentoo compilation emits warning:
      "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
 * ASTERISK-24225 - Dial option z is broken (Reported by
 * ASTERISK-24234 - app_meetme: Crash on conference shutdown due to
      NULL channel passed to meetme_stasis_generate_msg() (Reported by
      Shaun Ruffell)
 * ASTERISK-24043 - ARI /continue fails to actually continue into
      the dialplan (Reported by Krandon Bruse)
 * ASTERISK-24245 - gcc 4.1.2 complains of files that do not end
      with newlines (Reported by Shaun Ruffell)
 * ASTERISK-24229 - ARI: playback of sounds implicitly answers
      channel, preventing early media playback (Reported by Matt
 * ASTERISK-24178 - [patch]fromdomainport used even if not set
      (Reported by Elazar Broad)
 * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
      warnings and ref leaks (Reported by Walter Doekes)
 * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not
      be fully qualified domainname (Reported by Private Name)
 * ASTERISK-24147 - ARI: channel hangup crashes asterisk process
      (Reported by Edvin Vidmar)
 * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
      ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
 * ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to
      transmit ACK on received 200 OK (Reported by Aleksei Kulakov)
 * ASTERISK-24019 - When a Music On Hold stream starts it restarts
      at beginning of file. (Reported by Jason Richards)
 * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
      if ever not able to resolve (Reported by David Herselman)
 * ASTERISK-24264 - ARI: Adding a channel to a holding bridge
      automatically starts MOH (Reported by Samuel Galarneau)
 * ASTERISK-24212 - testsuite: Sporadic crash due to assert on
      stopping RTP engine (Reported by Matt Jordan)
 * ASTERISK-24241 - crash: CDRs recursively attempt to update Party
      B information in a multi-party bridge, overrunning the stack
      (Reported by Deepak Singh Rawat)
 * ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated
      during dial operation (Reported by Matt Jordan)
 * ASTERISK-24231 - crash: CLI execution of realtime destroy
      sippeers id 1 causes crash due to NULL name provided to
      ast_variable (Reported by Niklas Larsson)
 * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
 * ASTERISK-23577 - res_rtp_asterisk: Crash in
      ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
      Jay Jideliov)
 * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
      concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
      by Roman Skvirsky)
 * ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count of
      list items (Reported by Mark Michelson)
 * ASTERISK-24331 - Unexpected Errors in Asterisk Manager Interface
      Output (Reported by xrobau)
 * ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code when
      subscribing to an event with an unexpected body type (Reported
      by Mark Michelson)
 * ASTERISK-24301 - Security: Out of call MESSAGE requests
      processed via Message channel driver can crash Asterisk
      (Reported by Matt Jordan)
 * ASTERISK-24290 - Endpoint identifier match value fails to parse
      when CIDR network format is specified (Reported by Ray Crumrine)
 * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer.
      (Reported by Richard Mudgett)

Improvements made in this release:
 * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
      utility (Reported by Jeremy Lainé)

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

Thank you for your continued support of Asterisk!

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