[asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[Florian Wolters]

Matthew and list,

thanks for your detailed reply.

> This is a little hard to diagnose without seeing the SIP traffic for the
> duration of the call.  It makes it impossible to tell if the INVITES the
> provider is sending are related to the call (i.e. have the same Call-ID
> header),
> but if they are being sent consistently 15 minutes into every call it may
> not
> matter.  If the provider is sending you unsolicited INVITES that cause
> your
> calls to drop, I'd suggest contacting their customer service and asking
> them why
> they are being sent.

Does it make sense to have a more detailed tcpdump of the SIP session? If
so, how should such a thing been shared without posting too much ASCII
text to the list?

> The provider actually sent you two INVITES in rapid succession with
> different Call-IDs.

Sorry, but I have to give an update about this. After thinking about the
dump again, it dawned me. I set up a call forward back to my office phone
to test this issue. -.- Should have had a thought about that earlier.
Soorrryyy.

So I did setup another Extension leading me to a MeetMe conference to at
least listen to some MoH while waiting for the 15 Minutes to exceed. This
showed the same behaviour. After exactly 15 Minutes, the call is
terminated  - namely by the provider. The analysis of the dump in
Wireshark shows the last 6 SIP packets:

2013-03-21 15:56:50.648141    217.0.17.170   =>   172.16.0.2    Request:
INVITE sip:02341234567890 at 79.253.136.186:5060
2013-03-21 15:56:50.648325    172.16.0.2     =>   217.0.17.170  Status:
100 Trying
2013-03-21 15:56:50.648427    172.16.0.2     =>   217.0.17.170  Status:
200 OK, with session description
2013-03-21 15:56:50.731436    217.0.17.170   =>   172.16.0.2    Request:
ACK sip:02341234567890 at 79.253.136.186:5060
2013-03-21 15:56:50.735426    217.0.17.170   =>   172.16.0.2    Request:
BYE sip:02341234567890 at 79.253.136.186:5060
2013-03-21 15:56:50.735590    172.16.0.2     =>   217.0.17.170  Status:
200 OK

(manually copied that from the Wireshark window). This looks to me as if
the provider for some reason does an INVITE after 15 Minutes, that is not
correctly handled by my Asterisk. Is there any timer inside the SIP
protocol, that may be aged by 15 Minutes? Or should I have a deeper look
at the SIP packets?

Best regards

   Flo