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comms/asterisk16,
The Asterisk Software PBX
Branch: pkgsrc-2010Q4,
Version: 1.6.2.16.1,
Package name: asterisk-1.6.2.16.1,
Maintainer: jnemethAsterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
NOTE: This version does not work with the zaptel drivers. It
requires the newer DAHDI drivers which are still being ported.
So, there is no hardware support available at this moment.
Required to run:[
lang/perl5] [
www/curl] [
textproc/libxml2]
Required to build:[
devel/pkg-config] [
devel/gmake] [
devel/bison]
Package options: ldap
Master sites: (Expand)
Version history: (Expand)
- (2011-01-25) Updated to version: asterisk-1.6.2.16.1
- (2011-01-25) Package added to pkgsrc.se, version asterisk-1.6.2.15 (created)
CVS history: (Expand)
2011-01-25 13:43:16 by Matthias Scheler | Files touched by this commit (3) | |
Log message:
Pullup ticket #3335 - requested by gls
comms/asterisk16: security update
Revisions pulled up:
- comms/asterisk16/Makefile 1.28-1.29
- comms/asterisk16/distinfo 1.19-1.20
- comms/asterisk16/patches/patch-aq 1.10
---
Module Name: pkgsrc
Committed By: jnemeth
Date: Sun Jan 16 06:30:57 UTC 2011
Modified Files:
pkgsrc/comms/asterisk16: Makefile distinfo
pkgsrc/comms/asterisk16/patches: patch-aq
Log message:
Update to 1.6.2.16:
The release of Asterisk 1.6.2.16 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched,
tested
by russellb)
* Resolve issue where channel redirect function (CLI or AMI) hangs up
the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)
* Linux and *BSD disagree on the elements within the ucred structure. Detect
which one is in use on the system.
(Closes issue #18384. Reported, patched, tested by bjm, tilghman)
* app_followme: Don't create a Local channel if the target extension
does not
exist.
(Closes issue #18126. Reported, patched by junky)
* Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou.
rsw686.
Tested by cmbaker82)
* Resolve issue where REGISTER request with a Call-ID matching an existing
transaction is received it was possible that the REGISTER request would
overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16
---
Module Name: pkgsrc
Committed By: jnemeth
Date: Fri Jan 21 05:13:12 UTC 2011
Modified Files:
pkgsrc/comms/asterisk16: Makefile distinfo
Log message:
Update to 1.6.2.16.1
This is to fix AST-2011-001: Stack buffer overflow in SIP channel driver
Asterisk Project Security Advisory - AST-2011-001
Product Asterisk
Summary Stack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
Susceptibility Remote Authenticated Sessions
Severity Moderate
Exploits Known No
Reported On January 11, 2011
Reported By Matthew Nicholson
Posted On January 18, 2011
Last Updated On January 18, 2011
Advisory Contact Matthew Nicholson <mnicholson at digium.com>
CVE Name
Description When forming an outgoing SIP request while in pedantic
mode, a
stack buffer can be made to overflow if supplied with
carefully crafted caller ID information. This vulnerability
also affects the URIENCODE dialplan function and in some
versions of asterisk, the AGI dialplan application as well..
The ast_uri_encode function does not properly respect the
size
of its output buffer and can write past the end of it when
encoding URIs.
For full details, see:
http://downloads.digium.com/pub/security/AST-2011-001.html
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