Florian Wolters wrote: > > 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?
SIP sessions are generally small enough to post right to the list. Otherwise, you can put them up on a site like pastebin.com and provide the link. > 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@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@79.253.136.186:5060 > 2013-03-21 15:56:50.735426 217.0.17.170 => 172.16.0.2 Request: > BYE sip:02341234567890@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? This is where a full SIP trace that includes the messages used to setup the call in the first place would be helpful. I haven't seen anything related to session timers in what you've posted so far, but they may have been negotiated when the call was established. Regardless, your calls are consistently dropping at 15 minutes and you've shown that it's caused by the provider sending an INVITE, waiting for the OK, and then sending a BYE. You have enough to go to them and ask why it's happening. Even if it's something in your Asterisk configuration, they are initiating the hangup and should be able to tell you why. If they can't or won't help you troubleshoot this problem then I'd seriously consider looking for a new provider. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- 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