On Mar 22, 2013, at 5:22 AM, Florian Wolters <[email protected]> wrote: > > 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?
Sip session timers? http://doxygen.asterisk.org/trunk/sip_session_timers.html -d -- _____________________________________________________________________ -- 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
