[asterisk-users] Disconnect calls : known reasons

[Jonas Kellens]

On 04-10-12 19:50, Chad Wallace wrote:
> 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).

It is Asterisk that sends the BYE. I wouldn't know why. It has nothing 
to do with tunnels and so...



Jonas.