Hi,
Is it possible yet to restart a single Dahdi span in any version of
Asterisk? (instead of all of them)

Thanks.

Hi Carlos

It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk.
I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchange DR2 signalling between Nortel and Asterisk is about 5 sec.
Best regards

Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

Hi,

I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:

- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP divert is in place? If so, how?

- Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?

In both of the above cases, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?

Thanks,
Steve

I have an Asterisk server connected to a Nortel Pbx via an E1.  Everything works fine, I get calls in and out with callerid. The problem that has been reported to me is the following scenario:

A call comes in from the PSTN and is answered by Asterisk. The person dials the operator (1000) which is on the Nortel side so connection is made through the E1. The operator answers  and then transfers the call back to a SIP extension on the Asterisk (1303). The result is no audio and a dropped call.

My main theory at the moment is that when the receptionist hangs up after the transfer the E1 drops on the Nortel side. Anyone here with  this type of integration seen this problem?

Asterisk Project Security Advisory – AST-2012-006

Product Asterisk
Summary Remote Crash Vulnerability in SIP Channel Driver
Nature of Advisory Remote Crash
Susceptibility Remote Authenticated Sessions
Severity Moderate
Exploits Known No
Reported On April 16, 2012
Reported By Thomas Arimont
Posted On April 23, 2012
Last Updated On April 23, 2012
Advisory Contact Matt Jordan < mjordan AT digium DOT com >
CVE Name

Description A remotely exploitable crash vulnerability exists in the
SIP channel driver if a SIP UPDATE request is processed
within a particular window of time. For this to occur, the
following must take place:

1. The setting ‘trustrpid’ must be set to True

2. An UPDATE request must be received after a call has been
terminated and the associated channel object has been
destroyed, but before the SIP dialog associated with the
call has been destroyed. Receiving the UPDATE request
before the call is terminated or after the SIP dialog
associated with the call will not cause the crash
vulnerability described here.

3. The UPDATE request must be formatted with the
appropriate headers to reflect an Asterisk connected line
update. The information in the headers must reflect a
different Caller ID then what was previously associated
with the dialog.

When these conditions are true, Asterisk will attempt to
perform a connected line update with no associated channel,
and will crash.

Resolution Asterisk now ensures a channel exists before performing a
connected line update, when that connected line update is
initiated via a SIP UPDATE request.

In Asterisk versions not containing the fix for this issue,
setting the ‘trustrpid’ setting to False will prevent this
crash from occurring (default is False)

Affected Versions
Product Release Series
Asterisk Open Source 1.8.x All versions
Asterisk Open Source 10.x All versions
Asterisk Business Edition C.3.x All versions

Corrected In
Product Release
Asterisk Open Source 1.8.11.1, 10.3.1
Asterisk Business Edition C.3.7.4

Patches
SVN URL Revision
http://downloads.asterisk.org/pub/security/AST-2012-006-1.8.diff v1.8
http://downloads.asterisk.org/pub/security/AST-2012-006-10.diff v.10

Links https://issues.asterisk.org/jira/browse/ASTERISK-19770

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-2012-006.pdf and
http://downloads.digium.com/pub/security/AST-2012-006.html

Revision History
Date Editor Revisions Made
04/16/2012 Matt Jordan Initial release.

Asterisk Project Security Advisory – AST-2012-006
Copyright (c) 2012 Digium, Inc. All Rights Reserved.
Permission is hereby granted to distribute and publish this advisory in its
original, unaltered form.