----- Original Message ----- > From: "Howard Leadmon" <how...@leadmon.net> > To: asterisk-users@lists.digium.com > Sent: Saturday, November 24, 2012 3:19:10 PM > Subject: [asterisk-users] SIP Debugging Information.. > > > I did a little googling, but didn't seem to find anything specific > to > answer the question. I am trying to debug some calls on an Asterisk > system > (AsteriskNow) that are dropping, and when the general logs didn't > nail > anything I turned on SIP Debugging on the trunk to the provider. > Basically the complaint is that when some call in, regardless of if > the call > is answered, or if Vmail answers it, it drops the calls in a matter > of > seconds. The strange thing is, that the system processes many > hundreds of > calls daily, but only a couple specific incoming callers are seeing > the > drops. I would have thought a NAT issue, but why does this only > affect a > specific group of incoming callers, the rest go about their business > just > fine. I think thinking bandwidth.com is mucking something up, but > again I > have no specific proof one way or another, so why the debugging. > > When one of the problem callers is dropped, in the SIP debugging I > see: > > chan_sip.c: Scheduling destruction of SIP dialog > '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) > > > Is this the remote end (ie bandwidth.com) dropping the call, or is > the local > Asterisk server dropping the call?
[snip] > --- > [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: > <--- SIP read from UDP:216.82.224.202:5060 ---> > BYE sip:4104159270@10.98.4.36:5060 SIP/2.0 > Record-Route: <sip:216.82.224.202;lr;ftag=gK0b66d829> > Record-Route: <sip:67.231.4.93;lr=on;ftag=gK0b66d829> > Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKe902.53bde7e.0 > Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bKe902.32697e93.0 > Via: SIP/2.0/UDP 192.168.27.72:5060;branch=z9hG4bK0bBac8c2c3cb90659df > From: <sip:7173381800@192.168.27.72;isup-oli=0>;tag=gK0b66d829 > To: <sip:+14104159270@67.231.4.93>;tag=as0850c6db > Call-ID: 285991942_79966325@192.168.27.72 > CSeq: 297 BYE [snip] If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Michael (elguero) -- _____________________________________________________________________ -- 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
