[asterisk-users] Disconnect calls : known reasons

[Chad Wallace]

On Fri, 28 Sep 2012 11:03:05 +0200
Jonas Kellens <jonas.kellens at telenet.be> wrote:

> On 28-09-12 10:57, Administrator TOOTAI wrote:
> > Le 28/09/2012 10:40, Jonas Kellens a écrit :
> >> Maybe I need to explain a bit further : the call is send to the 
> >> IP-phone and answered. The call lasts for about 1 à 2 minutes and
> >> is then disconnected.
> >
> > We had this problem with some PSTN call termination providers, 
> > sometimes only against some destination.  I don't know if your 
> > incoming calls are 100% VOIP, I would start to see with providers.
> >
> > You may also check hangupcause and dialstatus variables.
> 
> Pure SIP.
> 
> Hangupcause 16
> 
> Dialstatus Answer
> 
> It has nothing to do with the provider-side.

You could narrow it down by inspecting the SIP packets for the call in
question (using wireshark or Asterisk sip debugging) and seeing which
end issues a BYE packet--if either one does.

Also, typically you only have contact with one end of a call (your
users) so it's very hard to say that something didn't happen on the
other end (somewhere out in the wild, where people drive through
tunnels).

PS, Sorry for the late reply... I haven't checked the list in a week.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0