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comms/asterisk18,
The Asterisk Software PBX
Branch: pkgsrc-2010Q4,
Version: 1.8.2.2,
Package name: asterisk-1.8.2.2,
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.
Asterisk 1.8 is a long term support version (i.e. it will be
supported for four years with an additional year of security only
fixes). See:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
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)
- (2011-01-26) Updated to version: asterisk-1.8.2.2
- (2011-01-25) Package added to pkgsrc.se, version asterisk-1.8.1.1 (created)
CVS history: (Expand)
2011-01-25 16:28:55 by Matthias Scheler | Files touched by this commit (3) | |
Log message:
Pullup ticket #3336 - requested by gls
comms/asterisk18: security update
Revisions pulled up:
- comms/asterisk18/Makefile 1.3-1.4
- comms/asterisk18/distinfo 1.5-1.6
- comms/asterisk18/patches/patch-aq 1.2
---
Module Name: pkgsrc
Committed By: jnemeth
Date: Sun Jan 16 17:52:43 UTC 2011
Modified Files:
pkgsrc/comms/asterisk18: Makefile distinfo
pkgsrc/comms/asterisk18/patches: patch-aq
Log message:
Update to 1.8.2:
The release of Asterisk 1.8.2 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:
* 'sip notify clear-mwi' needs terminating CRLF.
(Closes issue #18275. Reported, patched by klaus3000)
* Patch for deadlock from ordering issue between channel/queue locks in
app_queue (set_queue_variables).
(Closes issue #18031. Reported by rain. Patched by bbryant)
* 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)
* Fix reloading of peer when a user is requested. Prevent peer reloading
from
causing multiple MWI subscriptions to be created when using realtime.
(Closes issue #18342. Reported, patched by nivek.)
* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags
to 0
so res_jabber doesn't think there is already an XMPP connection sending
device state. Also clean up CLI commands a bit.
(Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
* Fixes issue with outbound google voice calls not working. Thanks to az1234
and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
---
odule Name: pkgsrc
Committed By: jnemeth
Date: Fri Jan 21 07:00:44 UTC 2011
Modified Files:
pkgsrc/comms/asterisk18: Makefile distinfo
Log message:
Update to 1.8.2.2
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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