[asterisk-users] Frequent Asterisk Restarts

[Mike Diehl]

On Thursday 10 November 2011 2:09:09 pm Danny Nicholas wrote:
> 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)

It's a phone-based transfer.

> 2. attended or blind transfer

The attended transfers seem to give them the most problems.  They say that if 
they hit the transfer softkey while the destination extension is still 
ringing, it tends to work well.


Mike.



> -----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.