[asterisk-users] SIP channel problem

[Joshua Colp]

----- Original Message -----
From: asterisk
[mailto:munkadolog at freemail.hu]
To: asterisk-users at lists.digium.com
Sent:
Mon, 31 Jul 2006 05:37:46 -0300
Subject: [asterisk-users] SIP channel
problem


> Hi,
> 
> We have a small callcenter.
> We registered with a provider with SIP with username and pass.
> If our customer start a call with an IP phone, he can speek.
> BUT
> If he hang Up the line, he get  a callback.
> with SIP debug I got the next message:
> 
> --- (24 headers 9 lines)---
> "Using INVITE request as basis request - ....."
> 
> 
> 
> 
> 
> Customer's PBX  -> call -> Provider's PBX -> PSTN
> If the customor give a Hang Up
> 
> Provider's PBX -> call -> Customer's PBX
> 
> I think.
> 
> I hope this is the situation because I can't find problem in the dial 
> plan. :)
> 
> Thx for help.
> Kind regards
> Szolke
> 

A full sip debug of the dialog would be very helpful so we would know who is at fault and what's going on.

Joshua Colp
Digium