Ops! Solo ora mi sono accorto che 3cx li ha comprati (quasi) tutti
Prima "PBX In a Flash" e poi "Elastix".
I 2 più importanti progetti basati su Asterisk e FreePBX GUI sono ormai belli e defunti.
Per quanto riguarda "PBX In a Flash" a quanto pare sono anche "spariti" i repositories mentre per "Elastix" dovrebbero rimamere (?) disponibili i repositories della 2.X e della 4.X.
Vedremo...
Asterweb
AST-2016-009: Remote unauthenticated sessions in chan_sip
Dal Team Asterisk Security (8 dicembre 2016).
Dal post originale:
Asterisk Project Security Advisory - ASTERISK-2016-009Product Asterisk
Summary
Nature of Advisory Authentication Bypass
Susceptibility Remote unauthenticated sessions
Severity Minor
Exploits Known No
Reported On October 3, 2016
Reported By Walter Doekes
Posted On
Last Updated On December 8, 2016
Advisory Contact Mmichelson AT digium DOT com
CVE NameDescription The chan_sip channel driver has a liberal definition for
whitespace when attempting to strip the content between a
SIP header name and a colon character. Rather than
following RFC 3261 and stripping only spaces and horizontal
tabs, Asterisk treats any non-printable ASCII character as
if it were whitespace. This means that headers such asContact\x01:
will be seen as a valid Contact header.
This mostly does not pose a problem until Asterisk is
placed in tandem with an authenticating SIP proxy. In such
a case, a crafty combination of valid and invalid To
headers can cause a proxy to allow an INVITE request into
Asterisk without authentication since it believes the
request is an in-dialog request. However, because of the
bug described above, the request will look like an
out-of-dialog request to Asterisk. Asterisk will then
process the request as a new call. The result is that
Asterisk can process calls from unvetted sources without
any authentication.If you do not use a proxy for authentication, then this
issue does not affect you.If your proxy is dialog-aware (meaning that the proxy keeps
track of what dialogs are currently valid), then this issue
does not affect you.If you use chan_pjsip instead of chan_sip, then this issue
does not affect you.Resolution chan_sip has been patched to only treat spaces and
horizontal tabs as whitespace following a header name. This
allows for Asterisk and authenticating proxies to view
requests the same wayAffected Versions
Product Release
Series
Asterisk Open Source 11.x All Releases
Asterisk Open Source 13.x All Releases
Asterisk Open Source 14.x All Releases
Certified Asterisk 13.8 All ReleasesCorrected In
Product Release
Asterisk Open Source 11.25.1, 13.13.1, 14.2.1
Certified Asterisk 11.6-cert16, 13.8-cert4Patches
SVN URL RevisionLinks
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/securityThis document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/ASTERISK-2016-009.pdf and
http://downloads.digium.com/pub/security/ASTERISK-2016-009.htmlRevision History
Date Editor Revisions Made
November 28, 2016 Mark Michelson Initial writeupAsterisk Project Security Advisory - ASTERISK-2016-009
Copyright (c) 2016 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
Dal Team Asterisk Security (8 dicembre 2016).
Dal post originale:
Asterisk Project Security Advisory - AST-2016-008Product Asterisk
Summary Crash on SDP offer or answer from endpoint using
Opus
Nature of Advisory Remote Crash
Susceptibility Remote unauthenticated sessions
Severity Critical
Exploits Known No
Reported On November 11, 2016
Reported By jorgen
Posted On
Last Updated On November 15, 2016
Advisory Contact jcolp AT digium DOT com
CVE NameDescription If an SDP offer or answer is received with the Opus codec
and with the format parameters separated using a space the
code responsible for parsing will recursively call itself
until it crashes. This occurs as the code does not properly
handle spaces separating the parameters. This does NOT
require the endpoint to have Opus configured in Asterisk.
This also does not require the endpoint to be
authenticated. If guest is enabled for chan_sip or
anonymous in chan_pjsip an SDP offer or answer is still
processed and the crash occurs.Resolution The code has been updated to properly handle spaces
separating parameters in the fmtp line. Upgrade to a
released version with the fix incorporated or apply patch.Affected Versions
Product Release
Series
Asterisk Open Source 13.x 13.12.0 and higher
Asterisk Open Source 14.x All VersionsCorrected In
Product Release
Asterisk Open Source 13.13.1, 14.2.1Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2016-008-13.diff Asterisk
13
http://downloads.asterisk.org/pub/security/AST-2016-008-14.diff Asterisk
14Links https://issues.asterisk.org/jira/browse/ASTERISK-26579
Asterisk Project Security Advisories are posted at
http://www.asterisk.org/securityThis document may be superseded by later versions; if so, the latest
version will be posted at
http://downloads.digium.com/pub/security/AST-2016-008.pdf and
http://downloads.digium.com/pub/security/AST-2016-008.htmlRevision History
Date Editor Revisions Made
November 15, 2016 Joshua Colp Initial draft of AdvisoryAsterisk Project Security Advisory - AST-2016-008
Copyright © 2016 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.
Rilasciato Asterisk 14.2.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.2.0.
Dal post originale:
The release of Asterisk 14.2.0 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett)
* ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configured (Reported by Joshua Colp)
* ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell)
New Features made in this release:
-----------------------------------
* ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events (Reported by Mark Michelson)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0
Rilasciato Asterisk 13.13.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.13.0.
Dal post originale:
The release of Asterisk 13.13.0 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:
New Features made in this release:
-----------------------------------
* ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter)
* ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud)
* ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud)
* ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell)
Improvements made in this release:
-----------------------------------
* ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0
Rilasciato Asterisk 11.25.0
Il giorno 23 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.25.0.
Dal post originale:
The release of Asterisk 11.25.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen)
* ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
Rilasciato Asterisk 14.2.0-rc2
Il giorno 22 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.2.0-rc2.
Dal post originale:
The release of Asterisk 14.2.0-rc2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0-rc2
Rilasciato Asterisk 13.13.0-rc2
Il giorno 22 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.13.0-rc2.
Dal post originale:
The release of Asterisk 13.13.0-rc2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0-rc2
Rilasciato Asterisk 13.12.2
Il giorno 10 novembre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.12.2.
Dal post originale:
The release of Asterisk 13.12.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2
Nuovo “Corso Asterisk 13 Avanzato” a Milano nei giorni 24-25-26 gennaio 2017
Nuovo "Corso Asterisk 13 Avanzato" a Milano nei giorni 24-25-26 gennaio 2017.
Sono aperte le iscrizioni al costo promozionale di € 390,00 più iva fino al 30/11/2016.
Vi aspettiamo.
Saluti, lo Staff
Foto di gruppo a fine “Corso Asterisk 13 Avanzato”
Un ringraziamento ed un "in bocca al lupo" ai partecipanti al corso.
Rilasciato Asterisk 14.1.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.1.1.
Dal post originale:
The release of Asterisk 14.1.1 resolves an issue reported by the community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1
Rilasciato Asterisk 13.12.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.12.1.
Dal post originale:
The release of Asterisk 13.12.1 resolves an issue reported by the community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1
Rilasciato Asterisk 11.24.1
Il giorno 27 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 11.24.1.
Dal post originale:
The release of Asterisk 11.24.1 resolves an issue reported by the community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
Rilasciato Asterisk 14.1.0
Il giorno 25 ottobre 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.1.0.
Dal post originale:
The release of Asterisk 14.1.0 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:
New Features made in this release:
-----------------------------------
* ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph)
* ASTERISK-26410 - core: Asterisk 14 doesn't show the header in the console or verbose when starting (Reported by Dan Jenkins)
* ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina)
* ASTERISK-26391 - Consoles do not display verbose logger messages even when requested. (Reported by Marcelo Terres)
* ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen)
* ASTERISK-26365 - rtp: Offer with multiple payloads for same codec is incorrectly handled (Reported by Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp)
* ASTERISK-26364 - res_pjsip: Don't assume a request will have target addresses (Reported by Joshua Colp)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft)
* ASTERISK-26341 - ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list (Reported by Matt Jordan)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on “core show channeltype Surrogate†in ast_format_cap_get_names (Reported by CGI.NET)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard)
* ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett)
* ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency, shouldn't be (Reported by Ben Merrills)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26283 - res_resolver_unbound: fails configure on older Ubuntu and CentOS (Reported by George Joseph)
* ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson)
* ASTERISK-26278 - asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. (Reported by Corey Farrell)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0