[asterisk-users] Frequent Asterisk Restarts

[Danny Nicholas]

You might want to see if it is
1. a phone or an asterisk transfer - phone transfer hits button on phone and
does attended/blind transfer that way; asterisk transfer initiated with *1
or #2 (whatever value is specified in features.conf)
2. attended or blind transfer.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike Diehl
Sent: Thursday, November 10, 2011 3:05 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Frequent Asterisk Restarts

I'll explore the options outlined in the document below, later tonight.

However, I've been able to reproduce the problem!  It seems that when one of
my users, at a particular site, transfers a call to another extension,
asterisk bounces.  

They're using Polycom 301's and 501's with SIP version 3.1.4.0070.

Without having gotten the debug info, yet, is there anything else I can look
at?

TIA.


On Thursday 10 November 2011 11:46:35 am Leif Madsen wrote:
> On 11-11-10 01:15 PM, Eric Wieling wrote:
> > The Asterisk source tree has a document with instructions on getting 
> > a backtrace from the segfaults so you can report it on the issue
tracker.
> 
> Most up to date documentation is on the Asterisk wiki:
> 
> https://wiki.asterisk.org/wiki/display/AST/Debugging
> 
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Take care and have fun,
Mike Diehl.

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