Download Asterisk 14.2.0 / 13.13.0 / 11.25.0

Asterisk is a comprehensive pbx for BSD, Linux and Mac OS X. The program offers all the functions you would expect from a telephone exchange. For example, it has options for voicemail, conferencing and call queuing. In addition, support for caller id services, adsi, sip and h323 is present. For a complete overview of all options, we refer to this page† The developers have released versions 14.2.0, 13.13.0, and 11.25.0 with the following announcements:

Asterisk 14.2.0 Now Available

Improvements made in this release:

• ASTERISK-26558 – app_queue: add variable to know if the call is not answered after a queue
• ASTERISK-26176 – chan_sip: Add AccountCode to AMI PeerEntry
• ASTERISK-26538 – codec_opus: Add sample to configs/samples/codecs.conf.sample
• ASTERISK-26488 – ARI: Add ‘ari show app’, ‘ari show apps’, and ‘ari set debug’ CLI commands
• ASTERISK-26418 – res_rtp_asterisk: Speed ​​up ICE resolution by blacklisting host subnets that are not involved in RTP

Bugs fixed in this release:

• ASTERISK-26608 – Compile and link failures on OpenBSD
• ASTERISK-26520 – codec_opus: Generated fmtp line has no content
• ASTERISK-26605 – codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
• ASTERISK-26516 – pjsip: Memory corruption with possible memory leak.
• ASTERISK-26556 – manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
• ASTERISK-26343 – ASTERISK-25951 causes issues for callerid manipulation through agi
• ASTERISK-26592 – Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
• ASTERISK-26565 – chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
• ASTERISK-26575 – test suite: Need to check PJSIP functionality when res_srtp is not loaded.
• ASTERISK-26571 – res_pjsip: Resolution incorrect when explicit IPv6 transport configuredASTERISK-26468 – ari: Bridge events stop working after this sequence of ARI calls
• ASTERISK-24400 – ooh323 sends wrong hangup code
• ASTERISK-26555 – Multi-party Video: Fix some post Asterisk-11 regressions
• ASTERISK-26412 – build: Prepare for gcc 6.2
• ASTERISK-26509 – A few non-critical deprecation warnings when building on Ubuntu 16.10
• ASTERISK-26523 – chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes – rtptimeout behaving badly – regression
• ASTERISK-26549 – app_dial: When PickupChan() is used some channels may have incorrect device state
• ASTERISK-24274 – [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
• ASTERISK-26311 – [patch] rtp_engine: Allow more than 32 dynamic payload types.
• ASTERISK-26506 – [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
• ASTERISK-25070 – Fix FTBFS on Hurd
• ASTERISK-26476 – chan_sip: Incorrect display option “Outbound reg. retry 403” in “sip show settings”
• ASTERISK-26541 – res_pjsip_sdp_rtp: Restrict number of formats to maximum
• ASTERISK-26537 – AMI: NewConnectedLine event is not documented
• ASTERISK-26526 – [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
• ASTERISK-26524 – astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
• ASTERISK-26344 – Asterisk 13.11.0 + PJSIP crash
• ASTERISK-26387 – Asterisk segfaults shortly after starting even with no active calls.
• ASTERISK-26513 – tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
• ASTERISK-26514 – Super Awesome Company: Don’t specify transport in pjsip.conf
• ASTERISK-26510 – pjproject_bundled uses the –strip-components option of tar which isn’t supported in older versions
• ASTERISK-22480 – Embedded pjproject: build.mak contains hardcoded full path to version.mak
• ASTERISK-26307 – res_pjsip_caller_id: Crash on outgoing change
• ASTERISK-26503 – app_voicemail: Asterisk crashes when MailboxExists is used
• ASTERISK-26423 – res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
• ASTERISK-26309 – [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
• ASTERISK-26482 – [patch] chan_pjsip: segfault on already disconnected session
• ASTERISK-26421 – Segmentation Fault with ARI originate into mixing bridge with 43 clients
• ASTERISK-26444 – ‘features show’ command in CLI does not return prompt.
• ASTERISK-26480 – [patch] CLI: core set debug: Auto-completes File not Module
• ASTERISK-26356 – menuselect: invalid test for GTK2
• ASTERISK-26462 – [patch] app_queue: While using queues with realtime, setting back to an empty context doesn’t stop the exit key usage
• ASTERISK-26439 – chan_rtp: Crash when originating
• ASTERISK-26457 – [patch] force_rport,auto_comedia: No NAT detection triggered.
• ASTERISK-26618 – build: Backport addition of librt check to configure.ac

New Features made in this release:

• ASTERISK-26595 – ARI: Add the ability to control the source of video in a multi-party mixing bridge
• ASTERISK-26492 – ARI: Add ability to specify channel variables on websocket events
• ASTERISK-26470 – ARI: Add an ‘asterisk_id’ field to outgoing events

For a full list of changes in this release, please see the ChangeLog:
/ChangeLog-14.2.0

Asterisk 13.13.0 Now Available

New Features made in this release:

• ASTERISK-26595 – ARI: Add the ability to control the source of video in a multi-party mixing bridge
• ASTERISK-26470 – ARI: Add an ‘asterisk_id’ field to outgoing events

Bugs fixed in this release:

• ASTERISK-26608 – Compile and link failures on OpenBSD
• ASTERISK-26343 – ASTERISK-25951 causes issues for callerid manipulation through agi
• ASTERISK-26520 – codec_opus: Generated fmtp line has no content
• ASTERISK-26605 – codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
• ASTERISK-26516 – pjsip: Memory corruption with possible memory leak.
• ASTERISK-26592 – Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
• ASTERISK-26565 – chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
• ASTERISK-26575 – test suite: Need to check PJSIP functionality when res_srtp is not loaded.
• ASTERISK-24400 – ooh323 sends wrong hangup code
• ASTERISK-26555 – Multi-party Video: Fix some post Asterisk-11 regressions
• ASTERISK-26412 – build: Prepare for gcc 6.2
• ASTERISK-26509 – A few non-critical deprecation warnings when building on Ubuntu 16.10
• ASTERISK-26523 – chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes – rtptimeout behaving badly – regression
• ASTERISK-26468 – ari: Bridge events stop working after this sequence of ARI calls
• ASTERISK-26311 – [patch] rtp_engine: Allow more than 32 dynamic payload types.
• ASTERISK-26549 – app_dial: When PickupChan() is used some channels may have incorrect device state
• ASTERISK-26541 – res_pjsip_sdp_rtp: Restrict number of formats to maximum
• ASTERISK-25070 – Fix FTBFS on Hurd
• ASTERISK-26476 – chan_sip: Incorrect display option “Outbound reg. retry 403” in “sip show settings”
• ASTERISK-26537 – AMI: NewConnectedLine event is not documented
• ASTERISK-26526 – [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
• ASTERISK-26524 – astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
• ASTERISK-26344 – Asterisk 13.11.0 + PJSIP crash
• ASTERISK-26387 – Asterisk segfaults shortly after starting even with no active calls.
• ASTERISK-26514 – Super Awesome Company: Don’t specify transport in pjsip.conf
• ASTERISK-26513 – tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
• ASTERISK-26510 – pjproject_bundled uses the –strip-components option of tar which isn’t supported in older versions
• ASTERISK-22480 – Embedded pjproject: build.mak contains hardcoded full path to version.mak
• ASTERISK-26307 – res_pjsip_caller_id: Crash on outgoing change
• ASTERISK-26503 – app_voicemail: Asterisk crashes when MailboxExists is used
• ASTERISK-26423 – res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
• ASTERISK-26309 – [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
• ASTERISK-26421 – Segmentation Fault with ARI originate into mixing bridge with 43 clients
• ASTERISK-26444 – ‘features show’ command in CLI does not return prompt.
• ASTERISK-26482 – [patch] chan_pjsip: segfault on already disconnected session
• ASTERISK-26480 – [patch] CLI: core set debug: Auto-completes File not Module
• ASTERISK-26356 – menuselect: invalid test for GTK2
• ASTERISK-26439 – chan_rtp: Crash when originating
• ASTERISK-26462 – [patch] app_queue: While using queues with realtime, setting back to an empty context doesn’t stop the exit key usage
• ASTERISK-26457 – [patch] force_rport,auto_comedia: No NAT detection triggered.
• ASTERISK-26618 – build: Backport addition of librt check to configure.ac

Improvements made in this release:

• ASTERISK-25063 – [patch]add X.509 subject alternative name support to Asterisk TLS support
• ASTERISK-26558 – app_queue: add variable to know if the call is not answered after a queue
• ASTERISK-26176 – chan_sip: Add AccountCode to AMI PeerEntry
• ASTERISK-26538 – codec_opus: Add sample to configs/samples/codecs.conf.sample
• ASTERISK-26488 – ARI: Add ‘ari show app’, ‘ari show apps’, and ‘ari set debug’ CLI commands
• ASTERISK-26418 – res_rtp_asterisk: Speed ​​up ICE resolution by blacklisting host subnets that are not involved in RTP

For a full list of changes in this release, please see the ChangeLog:
/ChangeLog-13.13.0

Asterisk 11.25.0 Now Available

Bugs fixed in this release:

• ASTERISK-26503 – app_voicemail: Asterisk crashes when MailboxExists is used
• ASTERISK-26480 – [patch] CLI: core set debug: Auto-completes File not Module
• ASTERISK-26356 – menuselect: invalid test for GTK2
• ASTERISK-26462 – [patch] app_queue: While using queues with realtime, setting back to an empty context doesn’t stop the exit key usage
• ASTERISK-26457 – [patch] force_rport,auto_comedia: No NAT detection triggered.

For a full list of changes in this release, please see the ChangeLog:
/ChangeLog-11.25.0
Thank you for your continued support of Asterisk!