The Asterisk Development Team has announced the release of Asterisk 10.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 10.3.0 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:

* — Fix potential buffer overrun and memory leak when executing “sip
show peers”
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)

* — Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)

* — Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)

* — Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)

* — Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)

* — push ‘outgoing’ flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 1.8.10.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.10.0 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:

* — Prevent outbound SIP NOTIFY packets from displaying a port of 0 —
(Closes issue ASTERISK-19430. Reported by Schmooze Com)

* — Include iLBC source code for distribution with Asterisk —
(Closes issue ASTERISK-18943. Reported by Leif Madsen)

* — Fix callerid of originated calls —
(Closes issue ASTERISK-19385. Reported by ornix)

* — Fix outbound DTMF for inband mode of chan_ooh323 —
(Closes issue ASTERISK-19233. Reported, patched by Matt Behrens)

* — Create and initialize udptl only when dialog requests image media —
(Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt)

* — Don’t prematurely stop SIP session timer —
(Closes issue ASTERISK-18996. Reported by Thomas Arimont)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.10.0

Thank you for your continued support of Asterisk!

The Asterisk Development Team has announced the release of Asterisk 10.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.2.0 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:

* — Prevent outbound SIP NOTIFY packets from displaying a port of 0 —
(Closes issue ASTERISK-19430. Reported by Schmooze Com)

* — Include iLBC source code for distribution with Asterisk —
(Closes issue ASTERISK-18943. Reported by Leif Madsen)

* — Fix callerid of originated calls —
(Closes issue ASTERISK-19385. Reported by ornix)

* — Fix outbound DTMF for inband mode of chan_ooh323 —
(Closes issue ASTERISK-19233. Reported, patched by Matt Behrens)

* — Create and initialize udptl only when dialog requests image media —
(Closes issue ASTERISK-16794. Reported by under, tested by Stefan Schmidt)

* — Don’t prematurely stop SIP session timer —
(Closes issue ASTERISK-18996. Reported by Thomas Arimont)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.2.0

Thank you for your continued support of Asterisk!

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 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:

* AST-2012-001: prevent crash when an SDP offer is received with an encrypted video stream when support for video is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
by: Matt Jordan

* Fix timing source dependency issues with MOH. Prior to this patch,
res_musiconhold existed at the same module priority level as the timing
sources that it depends on. This would cause a problem when music on
hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority
level, AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies.
(closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patched by elguero

* Fix RTP reference leak. If a blind transfer were initiated using a
REFER without a prior reINVITE to place the call on hold, AND if Asterisk
were sending RTCP reports, then there was a reference leak for the
RTP instance of the transferrer.
(closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an ‘h’ extension
is present. This prevents the ‘h’ extension from being run on the
transferee channel when it is transferred via a native transfer
mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
from chan_dahdi lost call progress detection functionality. Fix
analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller Patched by Richard Miller

* Fix regression that ‘rtp/rtcp set debup ip’ only works when a port
was also specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed 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-1.8.9.0

Thank you for your continued support of Asterisk!

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.

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!