ASTERWEB Blog

15Feb/17Off

Rilasciato Asterisk 14.3.0

Il giorno 13 febbraio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.3.0.

Dal post originale:

The release of Asterisk 14.3.0 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:

New Features made in this release:
-----------------------------------
* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft)
* ASTERISK-26767 - ARI channelvars cause memory leak (Reported by Sébastien Duthil)
* ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek)
* ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and
'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina)
* ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref count trap tripped. (Reported by Richard Mudgett)
* ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley)
* ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy)
* ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin)
* ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson)
* ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson)
* ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson)
* ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier)
* ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp)
* ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett)
* ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen)
* ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen)
* ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem)
* ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An)
* ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud)
* ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud)
* ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp)
* ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson)
* ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE)
* ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett)
* ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens)
* ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton)
* ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose)
* ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph)
* ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes)
* ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H)
* ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph)
* ASTERISK-26647 - Support older DNS style for OpenBSD (Reported by snuffy)
* ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis)
* ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi)
* ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari)
* ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka)
* ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen)
* ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion)
* ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron)
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy)

Improvements made in this release:
-----------------------------------
* ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton)
* ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav)
* ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi)
* ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0

15Gen/17Off

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

10Dic/16Off

AST-2016-009: Remote unauthenticated sessions in chan_sip

Dal Team Asterisk Security (8 dicembre 2016).

Dal post originale:

             Asterisk Project Security Advisory - ASTERISK-2016-009

Product 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 Name

Description 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 as

Contact\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 way

Affected 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 Releases

Corrected In
Product Release
Asterisk Open Source 11.25.1, 13.13.1, 14.2.1
Certified Asterisk 11.6-cert16, 13.8-cert4

Patches
SVN URL Revision

Links

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This 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.html

Revision History
Date Editor Revisions Made
November 28, 2016 Mark Michelson Initial writeup

Asterisk 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.

10Dic/16Off

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-2016-008

Product 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 Name

Description 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 Versions

Corrected In
Product Release
Asterisk Open Source 13.13.1, 14.2.1

Patches
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
14

Links https://issues.asterisk.org/jira/browse/ASTERISK-26579

Asterisk Project Security Advisories are posted at
http://www.asterisk.org/security

This 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.html

Revision History
Date Editor Revisions Made
November 15, 2016 Joshua Colp Initial draft of Advisory

Asterisk 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.

24Nov/16Off

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

24Nov/16Off

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

24Nov/16Off

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

24Nov/16Off

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

24Nov/16Off

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

12Nov/16Off

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

29Ott/16Off

Nuovo “Corso Asterisk 13 Avanzato” a Milano nei giorni 24-25-26 gennaio 2017

logo-asterweb

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

28Ott/16Off

Foto di gruppo a fine “Corso Asterisk 13 Avanzato”

Un ringraziamento ed un "in bocca al lupo" ai partecipanti al corso.

20161027-1024


20161027-27-1024


20161027-30-1024


20161027-31-1024

28Ott/16Off

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

28Ott/16Off

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

28Ott/16Off

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