Rilasciato Asterisk 12.6.0

Il giorno 24 settembre 2014, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 12.6.0.

Dal post originale:
The release of Asterisk 12.6.0 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:

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, please see the ChangeLog:

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