[asterisk-users] Dropping Conference calls

[Mark Rosedale]

So I didn't have sip debug set. So I don't have any SIP TIMER's in my log. I have that set now. 

I would be interested in the debut/logs if you have them.

I do have Spawn extension...exited non-zero on 'SIP/'

Here is the specifics 
VERBOSE[10928] pbx.c:   == Spawn extension (from-sip, 1***, 1) exited non-zero on 'SIP/7XXX-000009d7'

Not sure if that relates or not, but it is the only hit for the connection between my sip client and the PRI going outbound right before the hangup. 
On Jul 1, 2011, at 11:21 AM, Jonathan Thomas wrote:

> The key item in my logs, which would preface the call dropping, was: 
> [2011-06-28 09:43:49] DEBUG[25563] chan_sip.c: ** SIP TIMER: Cancelling
> retransmit of packet (reply received) Retransid #858
> 
> For instance - a call would be connected.  SIP debug/core debug on.  At the
> 14:30 mark I would begin tailing the full log.  Once I saw the SIP TIMER
> notice, it would be followed by a new INVITE (re-invite) SIP transmission
> that would be sent to the phone currently on call.  This re-invite was odd
> in that it would be on a different port to the phone than was already
> established (for example the NAT outgoing SIP OPTIONS would be sent to the
> phone on port 27608 - and this re-invite might go out on port 35780).  The
> behavior following would be: Asterisk would hang up as though the parties
> disconnected - however the phone would show the call was still going and
> would continue sending SIP responses to asterisk indicating as such.  When
> the phone was manually hung up it would send a SIP BYE (as normal) to
> asterisk - indicating it had no notice that Asterisk dropped the call.
> 
> Adding to sip.conf
> 	session-timers=refuse
> Resolved the issue by stopping Asterisk from sending these re-invites during
> a live call.
> 
> Hope that helps!  I have more SIP debugs/logs if they're useful to ya.
> 
> JT
> 
> 
> -----Original Message-----
> From: Mark Rosedale [mailto:mrosedale at oreilly.com] 
> Sent: Friday, July 01, 2011 10:45 AM
> To: jonathan.thomas at us.patersons.net; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [asterisk-users] Dropping Conference calls
> 
> What would I be looking for in the logs to indicate that time? 
> 
> I'm looking into the sip session timers. I believe the issue lies there, but
> haven't confirmed that just yet. 
> On Jul 1, 2011, at 10:31 AM, Jonathan Thomas wrote:
> 
>> 900ms?
> 
> 
> 
> 
> Email has been scanned for viruses

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