ASTERWEB Blog

17Set/17Off

AST-2017-008: RTP/RTCP information leak


Asterisk Project Security Advisory - AST-2017-008

Product Asterisk
Summary RTP/RTCP information leak
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known Yes
Reported On September 1, 2017
Reported By Klaus-Peter Junghanns
Posted On September 19, 2017
Last Updated On September 19, 2017
Advisory Contact Richard Mudgett
CVE Name CVE-2017-14099

Description This is a follow up advisory to AST-2017-005.

Insufficient RTCP packet validation could allow reading
stale buffer contents and when combined with the “nat” and
“symmetric_rtp” options allow redirecting where Asterisk
sends the next RTCP report.

The RTP stream qualification to learn the source address of
media always accepted the first RTP packet as the new
source and allowed what AST-2017-005 was mitigating. The
intent was to qualify a series of packets before accepting
the new source address.

Resolution The RTP/RTCP stack will now validate RTCP packets before
processing them. Packets failing validation are discarded.
RTP stream qualification now requires the intended series of
packets from the same address without seeing packets from a
different source address to accept a new source address.

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 11.6 All Releases
Certified Asterisk 13.13 All Releases

Corrected In
Product Release
Asterisk Open Source 11.25.3, 13.17.2, 14.6.2
Certified Asterisk 11.6-cert18, 13.13-cert6

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2017-008-11.diff Asterisk
11
http://downloads.asterisk.org/pub/security/AST-2017-008-13.diff Asterisk
13
http://downloads.asterisk.org/pub/security/AST-2017-008-14.diff Asterisk
14
http://downloads.asterisk.org/pub/security/AST-2017-008-11.6.diff Certified
Asterisk
11.6
http://downloads.asterisk.org/pub/security/AST-2017-008-13.13.diff Certified
Asterisk
13.13

Links https://issues.asterisk.org/jira/browse/ASTERISK-27274

https://issues.asterisk.org/jira/browse/ASTERISK-27252

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This document may be superseded by later versions; if so, the latest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-2017-008.pdf and
http://downloads.digium.com/pub/security/AST-2017-008.html

2Set/17Off

AST-2017-007: Remote Crash Vulerability in res_pjsip


Asterisk Project Security Advisory - AST-2017-007

Product Asterisk
Summary Remote Crash Vulerability in res_pjsip
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions
Severity Moderate
Exploits Known No
Reported On August 30, 2017
Reported By Ross Beer
Posted On
Last Updated On August 30, 2017
Advisory Contact George Joseph
CVE Name

Description A carefully crafted URI in a From, To or Contact header
could cause Asterisk to crash.

Resolution Patched pjsip_message_ip_updater to properly ignore the
trigger URI.

Affected Versions
Product Release Series
Asterisk Open Source 13.15.0
Asterisk Open Source 14.4.0

Corrected In
Product Release
Asterisk Open Source 13.17.1, 14.6.1

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2017-007-13.diff Asterisk
13
http://downloads.asterisk.org/pub/security/AST-2017-007-14.diff Asterisk
14

Links https://issues.asterisk.org/jira/browse/ASTERISK-27152

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/.pdf
and http://downloads.digium.com/pub/security/.html

2Set/17Off

AST-2017-006: Shell access command injection in app_minivm


Asterisk Project Security Advisory - AST-2017-006

Product Asterisk
Summary Shell access command injection in app_minivm
Nature of Advisory Unauthorized command execution
Susceptibility Remote Authenticated Sessions
Severity Moderate
Exploits Known No
Reported On July 1, 2017
Reported By Corey Farrell
Posted On
Last Updated On July 11, 2017
Advisory Contact Richard Mudgett
CVE Name

Description The app_minivm module has an “externnotify” program
configuration option that is executed by the MinivmNotify
dialplan application. The application uses the caller-id
name and number as part of a built string passed to the OS
shell for interpretation and execution. Since the caller-id
name and number can come from an untrusted source, a
crafted caller-id name or number allows an arbitrary shell
command injection.

Resolution Patched Asterisk’s app_minivm module to use a different
system call that passes argument strings in an array instead
of having the OS shell determine the application parameter
boundaries.

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 11.6 All releases
Certified Asterisk 13.13 All releases

Corrected In
Product Release
Asterisk Open Source 11.25.2, 13.17.1, 14.6.1
Certified Asterisk 11.6-cert17, 13.13-cert5

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2017-006-11.diff Asterisk
11
http://downloads.asterisk.org/pub/security/AST-2017-006-13.diff Asterisk
13
http://downloads.asterisk.org/pub/security/AST-2017-006-14.diff Asterisk
14
http://downloads.asterisk.org/pub/security/AST-2017-006-11.6.diff Certified
Asterisk
11.6
http://downloads.asterisk.org/pub/security/AST-2017-006-13.13.diff Certified
Asterisk
13.13

Links https://issues.asterisk.org/jira/browse/ASTERISK-27103

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-2017-006.pdf and
http://downloads.digium.com/pub/security/AST-2017-006.html

2Set/17Off

AST-2017-005: Media takeover in RTP stack


Asterisk Project Security Advisory - AST-2017-005

Product Asterisk
Summary Media takeover in RTP stack
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known No
Reported On May 17, 2017
Reported By Klaus-Peter Junghanns
Posted On
Last Updated On August 30, 2017
Advisory Contact Joshua Colp
CVE Name

Description The "strictrtp" option in rtp.conf enables a feature of the
RTP stack that learns the source address of media for a
session and drops any packets that do not originate from
the expected address. This option is enabled by default in
Asterisk 11 and above.

The "nat" and "rtp_symmetric" options for chan_sip and
chan_pjsip respectively enable symmetric RTP support in the
RTP stack. This uses the source address of incoming media
as the target address of any sent media. This option is not
enabled by default but is commonly enabled to handle
devices behind NAT.

A change was made to the strict RTP support in the RTP
stack to better tolerate late media when a reinvite occurs.
When combined with the symmetric RTP support this
introduced an avenue where media could be hijacked. Instead
of only learning a new address when expected the new code
allowed a new source address to be learned at all times.

If a flood of RTP traffic was received the strict RTP
support would allow the new address to provide media and
with symmetric RTP enabled outgoing traffic would be sent
to this new address, allowing the media to be hijacked.
Provided the attacker continued to send traffic they would
continue to receive traffic as well.

Resolution The RTP stack will now only learn a new source address if it
has been told to expect the address to change. The RTCP
support has now also been updated to drop RTCP reports that
are not regarding the RTP session currently in progress. The
strict RTP learning progress has also been improved to guard
against a flood of RTP packets attempting to take over the
media stream.

Affected Versions
Product Release
Series
Asterisk Open Source 11.x 11.4.0
Asterisk Open Source 13.x All Releases
Asterisk Open Source 14.x All Releases
Certified Asterisk 11.6 All Releases
Certified Asterisk 13.13 All Releases

Corrected In
Product Release
Asterisk Open Source 11.25.2, 13.17.1, 14.6.1
Certified Asterisk 11.6-cert17, 13.13-cert5

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2017-005-11.diff Asterisk
11
http://downloads.asterisk.org/pub/security/AST-2017-005-13.diff Asterisk
13
http://downloads.asterisk.org/pub/security/AST-2017-005-14.diff Asterisk
14
http://downloads.asterisk.org/pub/security/AST-2017-005-11.6.diff Certified
Asterisk
11.6
http://downloads.asterisk.org/pub/security/AST-2017-005-13.13.diff Certified
Asterisk
13.13

Links https://issues.asterisk.org/jira/browse/ASTERISK-27013

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-2017-005.pdf and
http://downloads.digium.com/pub/security/AST-2017-005.html

20Ago/17Off

Rilasciata beta di Asterisk 15 (no LTS)

Dal post di Matt Fredrickson:

It is with great pleasure I wish to inform the world of the first beta release of the new Asterisk 15 branch. It’s a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or so years. There has been a lot of work done in the Asterisk core to better support newer multi-stream video and WebRTC related technologies. For those who are interested, much of this will be covered in blog posts over the next month or two.

Typically, when a new major branch of Asterisk is created (13, 14, 15…), there are a few months of testing on the new branch that occurs prior to release in order to find regressions and other issues that may cause a first official release from the branch to be dead on arrival for a significant number of users. With today’s release of 15.0.0-beta1, this process has begun. Please feel free to start testing this version of Asterisk in as many adverse environments as possible. Any bugs should be reported on the Asterisk issue tracker at https://issues.asterisk.org/

As a side note, due to many of the core changes in the 15 branch that have been made since Asterisk 14 was released, it has been decided that Asterisk 15 will not be an LTS release. For those of you who are not familiar with the differences between LTS versus standard releases, you can find more information here.

Thanks to all the many Asterisk community members for providing so much help and support to make Asterisk the great open source project that it is.

Questo il link del post:
https://wiki.asterisk.org/wiki/display/AST/New+in+15

15Giu/17Off

Rilasciato Asterisk 14.6.0.

Il giorno 12 giugno 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.6.0.

Dal post originale:

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer)
* ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet)
* ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan)
* 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-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield)
* ASTERISK-25101 - DTLS configuration can not be specified in the general section - documentation (Reported by Ben Langfeld)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros )
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without locks (Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder)

Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton)
* ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0

15Giu/17Off

Rilasciato Asterisk 13.17.0

Il giorno 12 giugno 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.17.0.

Dal post originale:

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
* ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer)
* ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet)
* ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan)
* 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-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield)
* ASTERISK-25101 - DTLS configuration can not be specified in the general section - documentation (Reported by Ben Langfeld)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros )
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without locks (Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder)

Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton)
* ASTERISK-26976 - libsrtp-2.x.x support (Reported by Alex)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0

28Mag/17Off

AST-2017-004: Memory exhaustion on short SCCP packets

               Asterisk Project Security Advisory - AST-2017-004

Product Asterisk
Summary Memory exhaustion on short SCCP packets
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known No
Reported On April 13, 2017
Reported By Sandro Gauci
Posted On
Last Updated On April 13, 2017
Advisory Contact George Joseph <gjoseph AT digium DOT com>
CVE Name

Description A remote memory exhaustion can be triggered by sending an
SCCP packet to Asterisk system with “chan_skinny†enabled
that is larger than the length of the SCCP header but
smaller than the packet length specified in the header. The
loop that reads the rest of the packet doesn’t detect that
the call to read() returned end-of-file before the expected
number of bytes and continues infinitely. The “partial
data†message logging in that tight loop causes Asterisk to
exhaust all available memory.

Resolution If support for the SCCP protocol is not required, remove or
disable the module.

If support for SCCP is required, an upgrade to Asterisk will
be necessary.

Affected Versions
Product Release Series
Asterisk Open Source 11.x Unaffected
Asterisk Open Source 13.x All versions
Asterisk Open Source 14.x All versions
Certified Asterisk 13.13 All versions

Corrected In
Product Release
Asterisk Open Source 13.15.1, 14.4.1
Certified Asterisk 13.13-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/.pdf
and http://downloads.digium.com/pub/security/.html

Revision History
Date Editor Revisions Made
13 April 2017 George Joseph Initial report created

Asterisk Project Security Advisory -
Copyright © 2017 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

28Mag/17Off

AST-2017-003: Crash in PJSIP multi-part body parser

               Asterisk Project Security Advisory - AST-2017-003

Product Asterisk
Summary Crash in PJSIP multi-part body parser
Nature of Advisory Remote Crash
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known No
Reported On 13 April, 2017
Reported By Sandro Gauci
Posted On
Last Updated On April 13, 2017
Advisory Contact Mark Michelson <mark DOT michelson AT digium DOT
com>
CVE Name

Description The multi-part body parser in PJSIP contains a logical
error that can make certain multi-part body parts attempt
to read memory from outside the allowed boundaries. A
specially-crafted packet can trigger these invalid reads
and potentially induce a crash.

The issue is within the PJSIP project and not in Asterisk.
Therefore, the problem can be fixed without upgrading
Asterisk. However, we will be releasing a new version of
Asterisk where the bundled version of PJSIP has been
updated to have the bug patched.

If you are using Asterisk with chan_sip, this issue does
not affect you.

Resolution We have submitted the error report to the PJProject
maintainers and have coordinated a release...........

Affected Versions
Product Release
Series
Asterisk Open Source 11.x Unaffected
Asterisk Open Source 13.x All versions
Asterisk Open Source 14.x All versions
Certified Asterisk 13.13 All versions

Corrected In
Product Release
Asterisk Open Source 13.15.1, 14.4.1
Certified Asterisk 13.13-cert4

Patches
SVN URL Revision

Links https://issues.asterisk.org/jira/browse/ASTERISK-26939

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-2017-003.pdf and
http://downloads.digium.com/pub/security/AST-2017-003.html

Revision History
Date Editor Revisions Made
13 April, 2017 Mark Michelson Initial advisory created

Asterisk Project Security Advisory - AST-2017-003
Copyright (c) 2017 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

28Mag/17Off

AST-2017-002: Buffer Overrun in PJSIP transaction layer

               Asterisk Project Security Advisory - AST-2017-002

Product Asterisk
Summary Buffer Overrun in PJSIP transaction layer
Nature of Advisory Buffer Overrun/Crash
Susceptibility Remote Unauthenticated Sessions
Severity Critical
Exploits Known No
Reported On 12 April, 2017
Reported By Sandro Gauci
Posted On
Last Updated On April 13, 2017
Advisory Contact Mark Michelson <mark DOT michelson AT digium DOT
com>
CVE Name

Description A remote crash can be triggered by sending a SIP packet to
Asterisk with a specially crafted CSeq header and a Via
header with no branch parameter. The issue is that the
PJSIP RFC 2543 transaction key generation algorithm does
not allocate a large enough buffer. By overrunning the
buffer, the memory allocation table becomes corrupted,
leading to an eventual crash.

This issue is in PJSIP, and so the issue can be fixed
without performing an upgrade of Asterisk at all. However,
we are releasing a new version of Asterisk with the bundled
PJProject updated to include the fix.

If you are running Asterisk with chan_sip, this issue does
not affect you.

Resolution A patch created by the Asterisk team has been submitted and
accepted by the PJProject maintainers.

Affected Versions
Product Release
Series
Asterisk Open Source 11.x Unaffected
Asterisk Open Source 13.x All versions
Asterisk Open Source 14.x All versions
Certified Asterisk 13.13 All versions

Corrected In
Product Release
Asterisk Open Source 13.15.1, 14.4.1
Certified Asterisk 13.13-cert4

Patches
SVN URL Revision

Links https://issues.asterisk.org/jira/browse/ASTERISK-26938

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-2017-002.pdf and
http://downloads.digium.com/pub/security/AST-2017-002.html

Revision History
Date Editor Revisions Made
12 April, 2017 Mark Michelson Initial report created

Asterisk Project Security Advisory - AST-2017-002
Copyright (c) 2017 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.

25Mag/17Off

Rilasciato Asterisk 13.16.0-rc2

Il giorno 24 maggio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.16.0-rc2.

Dal post originale:

The following issues are resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------
[ASTERISK-26982] -
chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
(Reported by Stefan Engström)
[ASTERISK-26979] -
res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
(Reported by Javier Riveros )

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0-rc2

25Mag/17Off

Rilasciato Asterisk 14.5.0-rc2

Il giorno 24 maggio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.5.0-rc2.

Dal post originale:

The following issues are resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------
[ASTERISK-26982] -
chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
(Reported by Stefan Engström)
[ASTERISK-26979] -
res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
(Reported by Javier Riveros )

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0-rc2

23Mag/17Off

Rilasciato Asterisk 13.16.0-rc1

Il giorno 22 maggio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.16.0-rc1.

Dal post originale:

The following issues are resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------
[ASTERISK-25665] -
Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
[ASTERISK-26998] -
res_pjsip_session: INVITE retransmissions could still setup the same call again.
(Reported by Richard Mudgett)
[ASTERISK-26143] -
res_rtp_asterisk: One way audio when transcoding
(Reported by Henning Holtschneider)
[ASTERISK-26606] -
tcptls: Incorrect OpenSSL function call leads to misleading error report
(Reported by Bob Ham)
[ASTERISK-26983] -
Crash in Manager Reload when TLS Config Changes
(Reported by Joshua Elson)
[ASTERISK-25032] -
[patch]cel_odbc sometimes inserts CEL with wrong eventtime
(Reported by Etienne Lessard)
[ASTERISK-26173] -
func_cdr: CDR function does not permit empty values to be assigned
(Reported by gkloepfer)
[ASTERISK-25506] -
[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
(Reported by Frederic LE FOLL)
[ASTERISK-24529] -
Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
(Reported by Corey Farrell)
[ASTERISK-26860] -
Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
(Reported by Evers Lab)
[ASTERISK-26922] -
chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
[ASTERISK-26974] -
res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
[ASTERISK-26908] -
res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
(Reported by Richard Mudgett)
[ASTERISK-25823] -
SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Krüger)
[ASTERISK-26951] -
chan_sip: ACK with SDP does not update a direct media bridge
(Reported by Jean Aunis - Prescom)
[ASTERISK-26930] -
pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
(Reported by abelbeck)
[ASTERISK-26926] -
func_speex: Crash caused by frame with no datalen
(Reported by Richard Kenner)
[ASTERISK-26929] -
pjsip: Add database tables for RLS
(Reported by Joshua Colp)
[ASTERISK-26953] -
Asterisk crash if hep.conf have some missing parameters
(Reported by Joel Vandal)
[ASTERISK-26890] -
STUN server with non-default-route transport causes INVITE delay
(Reported by George Joseph)
[ASTERISK-26692] -
res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
(Reported by scgm11)
[ASTERISK-26835] -
res_rtp_asterisk: Crash when freeing RTCP address string
(Reported by Niklas Larsson)
[ASTERISK-26853] -
res_rtp_asterisk: Crash in pjnath when receiving packet
(Reported by Adagio)
[ASTERISK-26613] -
format_wav: wav16 format read file only by 320 - half of frame
(Reported by Vitaly K)
[ASTERISK-26169] -
format_ogg_vorbis: Memory leak using OGG in MixMonitor
(Reported by Ivan Myalkin)
[ASTERISK-21856] -
STUN never works when asterisk started without internet access
(Reported by Jeremy Kister)
[ASTERISK-20984] -
Audible clicks when playing sox encoded au file with STREAM FILE AGI command
(Reported by Roman S.)
[ASTERISK-26851] -
res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
[ASTERISK-26903] -
Listening TCP/TLS sockets stop when temporarily out of open files
(Reported by Walter Doekes)
[ASTERISK-26528] -
[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
(Reported by Badalian Vyacheslav)
[ASTERISK-26928] -
pjsip: Add database tables for PUBLISH support
(Reported by Joshua Colp)
[ASTERISK-26927] -
pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
(Reported by Alexander Traud)
[ASTERISK-26905] -
pjproject_bundled: Merge 3 upstream deadlock patches into bundled
(Reported by Ross Beer)
[ASTERISK-26897] -
chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
[ASTERISK-25974] -
Unused realtime MOH classes not purged on 'moh reload'
(Reported by Sébastien Couture)
[ASTERISK-26916] -
res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
[ASTERISK-21721] -
SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
[ASTERISK-26915] -
chan_sip: Session Timers required but refused wrongly.
(Reported by Alexander Traud)
[ASTERISK-26363] -
res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
(Reported by Yaacov Akiba Slama)
[ASTERISK-26896] -
Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
(Reported by twisted)
[ASTERISK-26705] -
libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
[ASTERISK-21009] -
xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
(Reported by Marcello Ceschia)
[ASTERISK-25490] -
[patch]SDP crypto tag is validated incorrectly
(Reported by Joerg Sonnenberger)
[ASTERISK-24712] -
xmpp: starttls problem causes connection spew
(Reported by Matthias Urlichs)
[ASTERISK-26086] -
res_musiconhold: format option is not documented adequately
(Reported by Jens Bürger)
[ASTERISK-23996] -
No core dumps because of res_musiconhold chdir.
(Reported by Walter Doekes)
[ASTERISK-26814] -
pjproject_bundled build fails to download pjproject source when using cURL
(Reported by Gergely Dömsödi)
[ASTERISK-23510] -
JABBER_STATUS fails with improper code 7 for unavailable clients
(Reported by Anthony Critelli)
[ASTERISK-21855] -
Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
(Reported by Jeremy Kister)
[ASTERISK-25622] -
WARNING for "JABBER: socket read error" should be more specific
(Reported by Sean Darcy)
[ASTERISK-26818] -
cdr: Problem setting variables in h exten
(Reported by scgm11)
[ASTERISK-26875] -
app_mixmonitor: Recording out of sync when 183 but no RTP
(Reported by Aaron An)

Improvements made in this release:
-----------------------------------

[ASTERISK-26088] -
Investigate heavy memory utilization by res_pjsip_pubsub
(Reported by Richard Mudgett)
[ASTERISK-26427] -
res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0-rc1

23Mag/17Off

Rilasciato Asterisk 14.5.0-rc1

Il giorno 22 maggio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.5.0-rc1.

Dal post originale:

The following issues are resolved in this release candidate:

Bugs fixed in this release:
-----------------------------------

[ASTERISK-25665] -
Duplicate logging in queue log for EXITEMPTY events
(Reported by Ove Aursand)
[ASTERISK-26998] -
res_pjsip_session: INVITE retransmissions could still setup the same call again.
(Reported by Richard Mudgett)
[ASTERISK-26143] -
res_rtp_asterisk: One way audio when transcoding
(Reported by Henning Holtschneider)
[ASTERISK-26606] -
tcptls: Incorrect OpenSSL function call leads to misleading error report
(Reported by Bob Ham)
[ASTERISK-26983] -
Crash in Manager Reload when TLS Config Changes
(Reported by Joshua Elson)
[ASTERISK-25032] -
[patch]cel_odbc sometimes inserts CEL with wrong eventtime
(Reported by Etienne Lessard)
[ASTERISK-26173] -
func_cdr: CDR function does not permit empty values to be assigned
(Reported by gkloepfer)
[ASTERISK-25506] -
[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
(Reported by Frederic LE FOLL)
[ASTERISK-24529] -
Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
(Reported by Corey Farrell)
[ASTERISK-26860] -
Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
(Reported by Evers Lab)
[ASTERISK-26922] -
chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
[ASTERISK-26974] -
res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
[ASTERISK-26908] -
res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
(Reported by Richard Mudgett)
[ASTERISK-25823] -
SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Krüger)
[ASTERISK-26951] -
chan_sip: ACK with SDP does not update a direct media bridge
(Reported by Jean Aunis - Prescom)
[ASTERISK-26930] -
pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
(Reported by abelbeck)
[ASTERISK-26926] -
func_speex: Crash caused by frame with no datalen
(Reported by Richard Kenner)
[ASTERISK-26929] -
pjsip: Add database tables for RLS
(Reported by Joshua Colp)
[ASTERISK-26953] -
Asterisk crash if hep.conf have some missing parameters
(Reported by Joel Vandal)
[ASTERISK-26890] -
STUN server with non-default-route transport causes INVITE delay
(Reported by George Joseph)
[ASTERISK-26692] -
res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
(Reported by scgm11)
[ASTERISK-26835] -
res_rtp_asterisk: Crash when freeing RTCP address string
(Reported by Niklas Larsson)
[ASTERISK-26853] -
res_rtp_asterisk: Crash in pjnath when receiving packet
(Reported by Adagio)
[ASTERISK-26613] -
format_wav: wav16 format read file only by 320 - half of frame
(Reported by Vitaly K)
[ASTERISK-26169] -
format_ogg_vorbis: Memory leak using OGG in MixMonitor
(Reported by Ivan Myalkin)
[ASTERISK-21856] -
STUN never works when asterisk started without internet access
(Reported by Jeremy Kister)
[ASTERISK-20984] -
Audible clicks when playing sox encoded au file with STREAM FILE AGI command
(Reported by Roman S.)
[ASTERISK-26851] -
res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
[ASTERISK-26903] -
Listening TCP/TLS sockets stop when temporarily out of open files
(Reported by Walter Doekes)
[ASTERISK-26528] -
[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
(Reported by Badalian Vyacheslav)
[ASTERISK-26928] -
pjsip: Add database tables for PUBLISH support
(Reported by Joshua Colp)
[ASTERISK-26927] -
pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
(Reported by Alexander Traud)
[ASTERISK-26905] -
pjproject_bundled: Merge 3 upstream deadlock patches into bundled
(Reported by Ross Beer)
[ASTERISK-26897] -
chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
[ASTERISK-25974] -
Unused realtime MOH classes not purged on 'moh reload'
(Reported by Sébastien Couture)
[ASTERISK-26916] -
res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
[ASTERISK-21721] -
SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
[ASTERISK-26915] -
chan_sip: Session Timers required but refused wrongly.
(Reported by Alexander Traud)
[ASTERISK-26363] -
res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
(Reported by Yaacov Akiba Slama)
[ASTERISK-26896] -
Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
(Reported by twisted)
[ASTERISK-26705] -
libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
[ASTERISK-21009] -
xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
(Reported by Marcello Ceschia)
[ASTERISK-25490] -
[patch]SDP crypto tag is validated incorrectly
(Reported by Joerg Sonnenberger)
[ASTERISK-24712] -
xmpp: starttls problem causes connection spew
(Reported by Matthias Urlichs)
[ASTERISK-26086] -
res_musiconhold: format option is not documented adequately
(Reported by Jens Bürger)
[ASTERISK-23996] -
No core dumps because of res_musiconhold chdir.
(Reported by Walter Doekes)
[ASTERISK-26814] -
pjproject_bundled build fails to download pjproject source when using cURL
(Reported by Gergely Dömsödi)
[ASTERISK-23510] -
JABBER_STATUS fails with improper code 7 for unavailable clients
(Reported by Anthony Critelli)
[ASTERISK-21855] -
Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
(Reported by Jeremy Kister)
[ASTERISK-25622] -
WARNING for "JABBER: socket read error" should be more specific
(Reported by Sean Darcy)
[ASTERISK-26818] -
cdr: Problem setting variables in h exten
(Reported by scgm11)
[ASTERISK-26875] -
app_mixmonitor: Recording out of sync when 183 but no RTP
(Reported by Aaron An)

Improvements made in this release:
-----------------------------------

[ASTERISK-26088] -
Investigate heavy memory utilization by res_pjsip_pubsub
(Reported by Richard Mudgett)
[ASTERISK-26427] -
res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0-rc1

13Apr/17Off

ISSABEL, il fork di Elastix per mantenere le versioni 2.5 e 4

ISSABEL è stato creato da diverse società con l'intento di mantenere, aggiornare e supportare tutti gli utenti di Elastix 2.5. e 4.

Questo il link del progetto: ISSABEL

ISSABEL, il fork di Elastix per mantenere le versioni 2.5 e 4