Asterisk News

Asterisk 12.0.0-alpha1 Now Available!

Aug 30, 2013

The Asterisk Development Team is pleased to announce the first alpha release of Asterisk 12.0.0.  This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the Asterisk 12 testing process.  Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. All Asterisk users are invited to participate in the #asterisk-bugs channel to help communicate issues found to the Asterisk developers. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list (http://lists.digium.com).

The first preliminary test release of Asterisk 12 is an alpha release, not a beta release. Due to the size and scope of the changes in Asterisk 12, both an alpha test cycle and a beta test cycle will be performed. While users are encouraged to participate in both test cycles, users who choose to participate in the alpha release testing should understand that an alpha release has not undergone all of the community testing that a beta release goes through.

Asterisk 12 is the next major release series of Asterisk. It will be a Standard release, similar to Asterisk 10.  For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 12, please see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12

A short list of some of the new major features includes:

  • A new SIP channel driver and accompanying SIP stack named chan_pjsip has been added. This new channel driver is based on the PJSIP SIP stack by Teluu. It includes support for the vast majority of features currently in chan_sip, as well as numerous architectural improvements that alleviate pain points present in the legacy SIP channel driver. Users who wish to use the new SIP channel driver are encouraged to read the instructions on installing and configuring PJSIP for Asterisk. Detailed instructions on configuring the new SIP stack in Asterisk can be found on the Asterisk wiki as wel. Test reports of successful use of chan_pjsip, with endpoint details, in addition to bug reports, are most welcome.

  • The Asterisk RESTful Interface (ARI) has been added. This interface lets external systems harness the telephony primitives within Asterisk to develop their own communications applications. Communication with Asterisk is done through a REST interface, while asynchronous events from Asterisk are encoded in JSON and sent via a WebSocket. More information on ARI can be found at https://wiki.asterisk.org/wiki/x/lYBbAQ.

  • Major standardization of the Asterisk Manager Interface and its events have occurred within this version. In particular, the names of Asterisk channels no longer change and are stable throughout the lifetime of the channel. More information on the changes in AMI can be seen in the AMI 1.4 Specification.

  • All bridging within Asterisk is now performed using the Asterisk Bridging API, which previously was only used by the ConfBridge application. This affords Asterisk users greater stability, and has resulted in the abstraction of channel masquerades, renaming, and other internal implementation details. It also allows for the seamless transition between two-party and multi-party bridges using core features.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Documentation

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

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.0.0-alpha1

Thank you for your continued support of Asterisk!

 

 


Asterisk 1.8.15-cert3, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, 11.2-cert2, and 11.5.1 Now Available (Security Release)

Aug 27, 2013

 

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, and 11.5.1.
 
These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

 
The release of these versions resolve the following issues:
  • A remotely exploitable crash vulnerability exists in the SIP channel driver if an ACK with SDP is received after the channel has been terminated. The handling code incorrectly assumes that the channel will always be present.
  • A remotely exploitable crash vulnerability exists in the SIP channel driver if an invalid SDP is sent in a SIP request that defines media descriptions before connection information. The handling code incorrectly attempts to reference the socket address information even though that information has not yet been set.                            
 
These issues and their resolutions are described in the security advisories.
 
For more information about the details of these vulnerabilities, please read security advisories AST-2013-004 and AST-2013-005, which were released at the same time as this announcement.
 
For a full list of changes in the current releases, please see the ChangeLogs:
The security advisories are available at:
Thank you for your continued support of Asterisk!

 


Asterisk 11.5.0 Now Available

Jul 15, 2013

The Asterisk Development Team has announced the release of Asterisk 11.5.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.5.0 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:

  • [ASTERISK-17386] - [patch] res_config_ldap with malloc_debug produces munmap_chunk(): invalid pointer:
  • [ASTERISK-17436] - random deadlocks - SIP messages not being processed
  • [ASTERISK-17458] - Deadlocks when using pthread timer
  • [ASTERISK-17467] - external moh is blocked when using dahdi timer
  • [ASTERISK-18207] - externnotify script called with (null) context parameter during pollmessages run, essentially stopping it from running.
  • [ASTERISK-19431] - Asterisk Russian language support missing voicemail prompts
  • [ASTERISK-19754] - Deadlock in chan_sip / pthread_timing
  • [ASTERISK-19883] - [patch] - RTP packet with Timestamp=0 on Multicast paging
  • [ASTERISK-20225] - Segmentation Fault on manager_play_dtmf sip_senddigit_end
  • [ASTERISK-20577] - Asterisk deadlocks waiting for timer in res_timing_pthread while running AGI script
  • [ASTERISK-21061] - Nortel I2004 unwanted autoanswer
  • [ASTERISK-21069] - xmpp distributed device states aggregation update fails
  • [ASTERISK-21120] - Unable to properly hang up calls when second line rings
  • [ASTERISK-21125] - Asterisk 11 needs libuuid in configure script due to pjproject
  • [ASTERISK-21151] - 'Squelching' early media in DAHDI (sig_pri)
  • [ASTERISK-21164] - Need clarification on distributed device state behavior and whether this behavior is a possible regression
  • [ASTERISK-21246] - [patch] use of rtpkeepalive uses CN packet with marker bit set, plus a ULAW payload instead of CN
  • [ASTERISK-21302] - [patch] app_voicemail crashes on config error and there are some potential memory leaks
  • [ASTERISK-21329] - chan_alsa: patch for crash when audio device in unexpected state
  • [ASTERISK-21356] - Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels
  • [ASTERISK-21374] - [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
  • [ASTERISK-21389] - res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely
  • [ASTERISK-21394] - [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
  • [ASTERISK-21397] - [patch] manager crash on unloading app_queue
  • [ASTERISK-21401] - [patch] codec_resample cannot be unloaded
  • [ASTERISK-21407] - [patch] features_shutdown doesn't finish cleanup
  • [ASTERISK-21409] - [patch] - Race condition with IAX2 transfer, 2 releases happen on same call legs. locks up with many threads blocked by iax2_destroy_helper
  • [ASTERISK-21412] - [patch] config.c/config_text_file_load() leaks globbuf
  • [ASTERISK-21429] - Distributed Device State using JABBER/XMPP not working since Secuity Advisory AST-2012-015
  • [ASTERISK-21430] - [patch] Call ID missing when logging through syslog
  • [ASTERISK-21466] - [patch] [crash] command (sip show peers) crashes Asterisk with ~3500 registered peers
  • [ASTERISK-21522] - [patch] DTMF end is not always processed, causes one-way audio
  • [ASTERISK-21664] - Asterisk terminates calls if Session-Expires isn't present on INVITE
  • [ASTERISK-21677] - NOTIFYs for BLF start queuing up and fail to be sent out
  • [ASTERISK-21716] - [patch] logger thread sometimes exits with messages still queued
  • [ASTERISK-21719] - [patch] res_srtp doesn't cleanup srtp library
  • [ASTERISK-21723] - [patch] pbx cleanup is incomplete
  • [ASTERISK-21724] - [patch] __ast_rwlock_destroy can segfault with DEBUG_THREADS
  • [ASTERISK-21738] - [patch] Segfault On Realtime Queue Members Processing
  • [ASTERISK-21742] - SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher.
  • [ASTERISK-21744] - [patch] - fix lower bound check with -ve integer conversion from a float
  • [ASTERISK-21779] - Manager closes connection when a SendText action is requested during hangup
  • [ASTERISK-21782] - Delayed audio to agent when answering a queue call
  • [ASTERISK-21785] - __ao2_ref_debug() logs to /tmp/refs when REF_DEBUG is not defined
  • [ASTERISK-21787] - No IAX2 communication either user/peer or friend accounts
  • [ASTERISK-21793] - Segmentation fault when dealing with Agent channels
  • [ASTERISK-21799] - [patch] Dropouts/distortion in MixMonitor recording when recording RTP with ptime of 60ms
  • [ASTERISK-21800] - ooh323 channels stuck if no gatekeer or ooh323 reload

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0

Thank you for your continued support of Asterisk!


Asterisk 1.8.23.0 Now Available

Jul 15, 2013

The Asterisk Development Team has announced the release of Asterisk 1.8.23.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.23.0 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:

  • [ASTERISK-17386] - [patch] res_config_ldap with malloc_debug produces munmap_chunk(): invalid pointer:
  • [ASTERISK-17436] - random deadlocks - SIP messages not being processed
  • [ASTERISK-17458] - Deadlocks when using pthread timer
  • [ASTERISK-17467] - external moh is blocked when using dahdi timer
  • [ASTERISK-18207] - externnotify script called with (null) context parameter during pollmessages run, essentially stopping it from running.
  • [ASTERISK-19431] - Asterisk Russian language support missing voicemail prompts
  • [ASTERISK-19754] - Deadlock in chan_sip / pthread_timing
  • [ASTERISK-19883] - [patch] - RTP packet with Timestamp=0 on Multicast paging
  • [ASTERISK-20225] - Segmentation Fault on manager_play_dtmf sip_senddigit_end
  • [ASTERISK-20577] - Asterisk deadlocks waiting for timer in res_timing_pthread while running AGI script
  • [ASTERISK-21069] - xmpp distributed device states aggregation update fails
  • [ASTERISK-21151] - 'Squelching' early media in DAHDI (sig_pri)
  • [ASTERISK-21164] - Need clarification on distributed device state behavior and whether this behavior is a possible regression
  • [ASTERISK-21225] - [patch] Setting nat=force_rport in [general] sip.conf will never work
  • [ASTERISK-21243] - [patch] Backport Appropiate NAT Setting Cleanups To 1.8
  • [ASTERISK-21246] - [patch] use of rtpkeepalive uses CN packet with marker bit set, plus a ULAW payload instead of CN
  • [ASTERISK-21302] - [patch] app_voicemail crashes on config error and there are some potential memory leaks
  • [ASTERISK-21329] - chan_alsa: patch for crash when audio device in unexpected state
  • [ASTERISK-21356] - Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels
  • [ASTERISK-21389] - res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely
  • [ASTERISK-21394] - [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present
  • [ASTERISK-21397] - [patch] manager crash on unloading app_queue
  • [ASTERISK-21407] - [patch] features_shutdown doesn't finish cleanup
  • [ASTERISK-21409] - [patch] - Race condition with IAX2 transfer, 2 releases happen on same call legs. locks up with many threads blocked by iax2_destroy_helper
  • [ASTERISK-21412] - [patch] config.c/config_text_file_load() leaks globbuf
  • [ASTERISK-21429] - Distributed Device State using JABBER/XMPP not working since Secuity Advisory AST-2012-015
  • [ASTERISK-21466] - [patch] [crash] command (sip show peers) crashes Asterisk with ~3500 registered peers
  • [ASTERISK-21522] - [patch] DTMF end is not always processed, causes one-way audio
  • [ASTERISK-21664] - Asterisk terminates calls if Session-Expires isn't present on INVITE
  • [ASTERISK-21677] - NOTIFYs for BLF start queuing up and fail to be sent out
  • [ASTERISK-21716] - [patch] logger thread sometimes exits with messages still queued
  • [ASTERISK-21719] - [patch] res_srtp doesn't cleanup srtp library
  • [ASTERISK-21723] - [patch] pbx cleanup is incomplete
  • [ASTERISK-21724] - [patch] __ast_rwlock_destroy can segfault with DEBUG_THREADS
  • [ASTERISK-21742] - SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher.
  • [ASTERISK-21744] - [patch] - fix lower bound check with -ve integer conversion from a float
  • [ASTERISK-21779] - Manager closes connection when a SendText action is requested during hangup
  • [ASTERISK-21782] - Delayed audio to agent when answering a queue call
  • [ASTERISK-21787] - No IAX2 communication either user/peer or friend accounts
  • [ASTERISK-21793] - Segmentation fault when dealing with Agent channels
  • [ASTERISK-21799] - [patch] Dropouts/distortion in MixMonitor recording when recording RTP with ptime of 60ms
  • [ASTERISK-21800] - ooh323 channels stuck if no gatekeer or ooh323 reload

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0

Thank you for your continued support of Asterisk!

 
 
 
 
                

Asterisk 11.5.0-rc1 Now Available

Jun 10, 2013

 

The Asterisk Development Team has announced the first release candidate of Asterisk 11.5.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
 
The release of Asterisk 11.5.0-rc1 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:
  • Fix a memory copying bug in slinfactory which was causing mixmonitor issues.
    (Closes issue ASTERISK-21799. Reported by Michael Walton)
  • app_voicemail: Add blank argument to externnotify if no context argument
    (Closes issue ASTERISK-18207. Reported by Barry L. Kline)
  • Fix CDR not being created during an externally initiated blind transfer
    (Closes issue ASTERISK-21394. Reported by Ishfaq Malik)
  • Fix One-Way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
    (Closes issue ASTERISK-21374. Reported by Michael L. Young)
  • Fix crash when AMI redirect action redirects two channels out of a bridge.
    (Closes issue ASTERISK-21356. Reported by William luke)
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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