ckleea

The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 10.0.0. This release candidate is available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk/

All Asterisk users are encouraged to participate in the Asterisk 10 testing
process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/jira. It is also very useful to see successful test
reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the Asterisk
versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
(More information available at
  https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

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

http://downloads.asterisk.org/pu ... hangeLog-10.0.0-rc1

Thank you for your continued support of Asterisk!

ckleea

The Asterisk Development Team has announced security releases for Asterisk 1.4,
1.6.2 and 1.8. The available security releases are released as versions 1.4.43,
1.6.2.21 and 1.8.7.2.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk versions 1.4.43, 1.6.2.21, and 1.8.7.2 resolves an issue
with possible remote enumeration of SIP endpoints with differing NAT settings.

The release of Asterisk versions 1.6.2.21 and 1.8.7.2 resolves a remote crash
possibility with SIP when the "automon" feature is enabled.

The issues and resolutions are described in the AST-2011-013 and AST-2011-014
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-013 and AST-2011-014, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pu ... es/ChangeLog-1.4.43
http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.21
http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.7.2

Security advisory AST-2011-013 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Security advisory AST-2011-014 is available at:

* http://downloads.asterisk.org/pub/security/AST-2011-013.pdf

Thank you for your continued support of Asterisk!

ckleea

The Asterisk Development Team has announced the third release candidate of
Asterisk 10.0.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.0.0-rc3 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 candidate:

* Add ASTSBINDIR to the list of configurable paths

This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.

* 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.

* Fix a change in behavior in 'database show' from 1.8.

In 1.8 and previous versions, one could use any fullword portion of
the key name, including the full key, to obtain the record. Until this
patch, this did not work for the full key.

* 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
1.6.2, this would mean if one setting was nat=yes or nat=route and the other
was either nat=no or nat=never. In 1.8 and 10, this would mean when one
was nat=force_rport and the other was nat=no.

In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all
possible.

* Fixed SendMessage stripping extension from To: header in SIP MESSAGE

When using the MessageSend application to send a SIP MESSAGE to a
non-peer, chan_sip stripped off the extension and failed to add it back
to the sip_pvt structure before transmitting. This patch adds the full
URI passed in from the message core to the sip_pvt structure.

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

http://downloads.asterisk.org/pu ... hangeLog-10.0.0-rc3

Thank you for your continued support of Asterisk!

bubblestar

Upated to Asterisk 1.8.7.2.  Thanks!

For Asterisk 10.0.0, I am looking forward to the official release other than the version of release candidate.  Once they are ready, I may try it together with CentOS 6.0 or 6.1

ckleea

回復 184# bubblestar

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/pu ... k/ChangeLog-1.8.8.0

Thank you for your continued support of Asterisk!

ckleea

The Asterisk Development Team is proud to announce the release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more information about
support time lines for Asterisk releases, see the Asterisk versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the version number per the blog post available at


http://blogs.digium.com/2011/07/ ... ved-at-asterisk-10/

The release of Asterisk 10 would not have been possible without the support and
contributions of the community.

You can find an overview of the work involved with the 10.0.0 release in the
summary:

http://svn.asterisk.org/svn/aste ... -10.0.0-summary.txt

A short list of available features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
associated with an active call can now be routed through the Asterisk
dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable of mixing
audio at sample rates ranging from 8kHz-192kHz
* Addition of video_mode option in confbridge.conf to provide basic video
conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES

Also, when upgrading a system between major versions, it is imperative that you
read and understand the contents of the UPGRADE.txt file, which is located at:

http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt

Thank you for your continued support of Asterisk!

ckleea

The Asterisk Development Team has announced the release of Asterisk 1.8.8.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.8.1 resolves a regression introduced in Asterisk
1.8.8.0 reported by the community, and would have not been possible without your
participation.  Thank you!

The following is the issue resolved in this release:

* 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, which may include having Music On Hold failing during a SIP Hold.

 (closes issue ASTERISK-19095)
 Reported by: Stefan Schmidt

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

http://downloads.asterisk.org/pu ... k/ChangeLog-1.8.8.1

Thank you for your continued support of Asterisk!

ckleea

The Asterisk Development Team is pleased to announce the first
release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.

2.6.0 is a feature release which:

  - Adds support for the TE820 8-span card to the wct4xxp driver.

  - Decrease load time of analog cards supported by the wctdm24xxp
    driver.

  - Adds sysfs object model to facilitate persistent span numbering
    and early loading of modules (NOTE: by default this release
    still behaves like previous releases with regards to span
    numbering assignment).

  - dahdi_pcap tool is now included in DAHDI-tools but not compiled
    by default since it depends on a currently unsupported interface
    in DAHDI-Linux. It is intended that in future releases this will
    be compiled by default.

Issues closed in this release:

DAHTOOL-49: adding pcap support to Dahdi
            (Reported by: Torrey Searle)
DAHLIN-258: weird sound with a native bridged isdn-bri connection
               (Reported by: Daniel)
DAHLIN-264: xpp: E1 CAS multiframe bits not properly set

DAHDI-Linux 2.6.0, DAHDI-Tools 2.6.0, and DAHDI-Linux-Complete
2.6.0+2.6.0 are available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

The DAHDI-Linux shortlog of changes that are not in 2.5.0.2:

Doug Bailey:
        wctdm24xxp, wcte12xp: Update VPMOCT032 firmware to 1.12.0.

Tzafrir Cohen:
        Avoid building PCI devices if kernel has no PCI
        xpp: Allow up to 128 Astribanks on a system
        xpp: increase command queue length to 1500
        xpp: USB_FW rev 10085: fix regression from r10013
        xpp: PIC_TYPE_1 rev 9841: followup to r10013
        bugfix: off-by-one in span assignment
        xpp: USB firmware r9964: minor bugfixes
        xpp: bugfix: clear NOTOPEN span alarm on assign
        xpp: bugfix -- manage xpd refcount for EC module
        xpp: Adaptations for E-Main-3
        xpp: remove leftovers of old XPD_STATE method
        README: Minor additions regarding pinned-spans
        README: initial update for span assignments
        dahdi: Add error messages in dahdi_ioctl_chanconfig.
        xpp: fix FXS D DTMF detection (not zero)
        xpp: fix bashism in xpp_debug
        live_dahdi: optionally generate FreePBX DB entries

Matthew Fredrickson:
        wct4xxp: Add support for TE820 and VPMOCT256.

Russ Meyerriecks:
        wct4xxp: Remove vpm400 support.
        wct4xxp: Revise vpm struct due to product name changes
        wct4xxp: Handle incorrect vpm module/card pairings
        wct4xxp: minor: Removed unnecessary instrumentation
        wct4xxp: Expose serial number in dahdi_device and kernel log.
        wct4xxp: Add field upgradable firmware support for TE820.
        wcte12xp, wctdm24xxp: Remove frowny face from vpmoct032 error message

Oron Peled:
        xpp: BRI: batch D-Channel packets to fix frag.
        xpp: BRI: split multibyte functionality
        xpp: BRI: remove trivial BRISTUFF wrappers
        xpp: BRI: remove legacy BRISTUFF code
        xpp: bad module_put() when too many Astribanks
        DAHDI-linux: Fix "surprise removal" problems
        xpp: BRI: fix timing priority calculation
        xpp: FXS: mwi and search_fsk fixes
        xpp: PRI: restore pri_protocol to R/W:
        xpp: pri: fix RS1 init in E1 CAS mode
        xpp: fxs: demote SETPOLARITY message to DBG()
        xpp: silence some bad ioctl() reporting
        xpp: restore backward compat dahdi_registration
        Extra debugging aids and messages
        xpp: cleanup some printk()'s
        added 'basechan' and 'channels' attributes to spans
        dahdi: Give userspace a chance to respond to surprise removal.
        xpp: Remove dahdi_autoreg parameter:
        xpp: more informative span description:
        xpp: make unregistration safer (idempotent)
        xpp: adapt to 'location' attribute removal:
        xpp: PRI: use DAHDI new set_spantype() method
        dahdi: Expose spans in sysfs.
        dahdi: dahdi_is_analog_span() -> dahdi_is_digital_span()
        dahdi: start handling "surprise device removal".

Shaun Ruffell:
        wctdm24xxp: Fix bug if hook state on FXS changes before channel configuration.
        wct4xxp: Reduce time spent waiting for auth done bit on TE820.
        wct4xxp: Fail startup if not generating interrupts.
        dahdi: Return dahdi_span_ops.startup callback errors to userspace.
        wctdm24xxp: Do not call voicebus_release() before wctdm_back_out_gracefully()
        dahdi: #include <linux/module.h> in dahdi/kernel.h and GpakCust.h
        wctc4xxp: Replace 'ndo_set_multicast_list' with 'set_rx_mode'
        wctdm24xxp: Wait for background threads to complete on failed load.
        dahdi: Unregister dahdi_device from sysfs if we fail to auto assign spans.
        dahdi: Fix typo in previous commit which forced some spans to always fail assignment.
        dahdi: First span registered becomes master by default.
        dahdi: Define POLLRDHUP on kernels < 2.6.17
        wctdm24xxp, wcte12xp: Allow VPMADT032 commands more time to complete.
        wct4xxp: Allow linemode (T1/E1/J1) to be changed via sysfs attribute.
        wcte12xp: Allow linemode (T1/E1) to be changed via sysfs attribute.
        dahdi: Allow 'spantype' to be changed before span assignement via sysfs.
        dahdi: Remove dahdi_span.irq and move dahdi_span.irqmisses into dahdi_device.
        dahdi: Expose dahdi devices in sysfs.
        dahdi: Register devices instead of individual spans.
        wct4xxp: Deprecate 't1e1override' module parameter in favor of 'default_linemode'.
        wct4xxp: Refactor t4_serial_setup() to remove t4.globalconfig.
        wct4xxp: Trivial. Use ARRAY_SIZE in free_wc() and __handle_leds().
        wct4xxp: Atomically perform some read/modify/write operations
        wct4xxp: Fix spelling
        wct4xxp: Change t4_span.spantype to linemode.
        wct4xxp: Trivial refactoring in t4_init_one().
        wct4xxp: Add has_e1_span() helper.
        wcte12xp: Deprecate 't1e1override' module parameter in favor of 'default_linemode'.
        wct4xxp: Remove redundant 'vpm' from struct t4.
        wct4xxp: Remove unused debugging code
        wct4xxp: Turn off the fancy alarm LEDS.
        wct4xxp: Hold a pointer to the devtype directly
        wct4xxp: Remove unused fields from 'struct t4' and 'struct t4_span'
        wct4xxp: Remove prefetching support.
        wct4xxp: Use in-hardirq version of dahdi_receive/transmit.
        wct4xxp: __t4_framer_in and __t4_framer_out speedups.
        wct4xxp: Remove 'pedanticpci' module parameter.
        wct4xxp: Remove some debug information from the kernel logs.
        wct4xxp: Slow down the rate we poll the framers in alarm.
        wct4xxp: Move "hardware DTMF disabled" message from dev_notice -> dev_info
        wctdm24xxp: Remove DEBOUNCING_RINGING_OFF from ring_detector_state enum.
        wctdm24xxp: Setup all VPMADT032 channels on hybrid cards.
        dahdi: Add functions for determining spantype (E1/T1) to include/dahdi/kernel.h
        dahdi: Define pr_xxx macros if not already defined.
        wcte12xp: Set uncollected performance counters to -1.
        wct4xxp: Remove unused t4_span.psync and t4_span.redalarms.
        wctdm24xxp: Remove fwringdetect module parameter.
        wctdm24xxp: Use interval for debouncing FXO polarity detection.
        wctdm24xxp: Use interval for debouncing FXO battery.
        wctdm24xxp: Use time interval for debouncing FXO ring detect.
        wctdm24xxp: Use interval for checking FXS on hook transfer timer.
        wctdm24xxp: 'oppending_ms' shouldn't assume being checked at 1ms intervals.
        wctdm24xxp: Name the shadow registers for each modules.
        wctdm24xxp: Change intcount to framecount.
        wctdm24xxp: Use fact that handle_transmit/receive are called in hard-irq
        wctdm24xxp: Protect creation / destruction of VPM instance.
        wctdm24xxp: Reset the polarity debounce setting when battery is lost.
        wctdm24xxp: Probe for and configure modules in parallel.
        wctdm24xxp: Introduce bg_create/bg_join.
        wcte12xp: Abort driver bind if read/write test fails.

The DAHDI-Linux diffstat from the 2.5.0.2 release:


For a full list of changes in these releases, please see the ChangeLog at
http://svn.asterisk.org/svn/dahdi/linux/tags/2.6.0/ChangeLog and
http://svn.asterisk.org/svn/dahdi/tools/tags/2.6.0/ChangeLog .

Issues found in this release can be reported in the DAHDI-Linux [1] and
DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN
[2] https://issues.asterisk.org/jira/browse/DAHTOOL

Thank you for your continued support of Asterisk!

bubblestar

Both Asterisk 1.8.8.1 and Dahdi 2.6.0 are updated and running without problem.  Thanks.

ckleea

I  just check that 1.8.9.0 and 10.1.0 are on RC1.
It seems there are so many bugs fixes. As far as I read, when one problem fixes, other may arise. This can be related to commercial VOIP phone functionalities.

bubblestar

Asterisk 1.8.8.2 released on 19-1-2012.  My update is in progress.

角色

How were the progress and result?

YH

ckleea

I have not done any update for quite a while

ckleea

The Asterisk Development Team is pleased to announce the release of
Asterisk 10.1.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.0 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

* Allow playback of formats that don't support seeking.  ast_streamfile
 previously did unconditional seeking on files that broke playback of
 formats that don't support that functionality.  This patch avoids the
 seek that was causing the problem.  
 (closes issue ASTERISK-18994) Patched by: Timo Teras

* Add pjmedia probation concepts to res_rtp_asterisk's learning mode.  In
 order to better handle RTP sources with strictrtp enabled (which is the
 default setting in 10) using the learning mode to figure out new sources
 when they change is handled by checking for a number of consecutive (by
 sequence number) packets received to an rtp struct based on a new
 configurable value called 'probation'.  Also, during learning mode instead
 of liberally accepting all packets received, we now reject packets until a
 clear source has been determined.

* 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/pu ... sk/ChangeLog-10.1.0

Thank you for your continued support of Asterisk!

ckleea

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/pu ... k/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!