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:[email protected]: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:[email protected]:5060 2013-03-21 15:56:50.735426 217.0.17.170 => 172.16.0.2 Request: BYE sip:[email protected]: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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
