[asterisk-dev] Asterisk 1.8.8.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the release of
Asterisk 1.8.8.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Updated SIP 484 handling; added Incomplete control frame
   When a SIP phone uses the dial application and receives a 484 Address
   Incomplete response, if overlapped dialing is enabled for SIP, then 
the 484
   Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE
   channel variable is set to 28. Previously, the Incomplete application
   dialplan logic was automatically triggered; now, explicit dialplan 
usage of
   the application is required.
     (Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew
     Jordan Review: https://reviewboard.asterisk.org/r/1416/)

* Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support 
IPv6
   and getting such addresses from DNS can cause error messages on the 
remote
   end involving bad IPv4 address casts in the presence of IPv6/IPv4 
tunnels.
     (Closes issue ASTERISK-18090. Patched by Kinsey Moore)

* Fix bad RTP media bridges in directmedia calls on peers separated by 
multiple
   Asterisk nodes.
     (Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue
     ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose)

* Fix crashes in ast_rtcp_write()
     (Closes issue ASTERISK-18570)
     Related issues that look like they are the same problem:
       (Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, 
ASTERISK-13334,
       ASTERISK-9977, ASTERISK-9716)
     Review: https://reviewboard.asterisk.org/r/1444/
     Patched by: Russell Bryant

* Fix for incorrect voicemail duration in external notifications.
   This patch fixes an issue where the voicemail duration was being reported
   with a duration significantly less than the actual sound file duration.
     (Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad 
House,
     Karsten Wemheuer, KevinH Tested by: Matt Jordan
     Review: https://reviewboard.asterisk.org/r/1443)

* Prevent segfault if call arrives before Asterisk is fully booted.
     (Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/)

* Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
     http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

* Fix locking order in app_queue.c which caused deadlocks
     (Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by 
Gregory
     Nietsky)
     (Closes issue ASTERISK-18487. Reported by Jason Legault, patched by 
Gregory
     Nietsky)

* Fix regression in configure script for libpri capability checks
     (Closes issue ASTERISK-18687. Reported by norbert, patched by Richard
     Mudgett)

* Prevent BLF subscriptions from causing deadlocks.
     (Closes issue ASTERISK-18663)
     Review: https://reviewboard.asterisk.org/r/1563/

* Fix deadlock if peer is destroyed while sending MWI notice.
     (Closes issue ASTERISK-18747)
     Reported by: Gregory Hinton Nietsky

* Fix issue with setting defaultenabled on categories that are already 
enabled
   by default.
     (Closes issue ASTERISK-18738)
     Reported by: Paul Belanger

* Don't crash on INFO automon request with no channel
     AST-2011-014. When automon was enabled in features.conf, it was 
possible
     to crash Asterisk by sending an INFO request if no channel had been
     created yet.

* Fixed crash from orphaned MWI subscriptions in chan_sip
   This patch resolves the issue where MWI subscriptions are orphaned
   by subsequent SIP SUBSCRIBE messages.

* Default to nat=yes; warn when nat in general and peer differ
     AST-2011-013. It is possible to enumerate SIP usernames when the 
general and
     user/peer nat settings differ in whether to respond to the port a 
request is
     sent from or the port listed for responses in the Via header. In 
1.4 and

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0

Thank you for your continued support of Asterisk!