[asterisk-users] SIP Debugging Information..

[Michael L. Young]

----- Original Message -----
> From: "Howard Leadmon" <howard at leadmon.net>
> To: asterisk-users at 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 at 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 at 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 at 192.168.27.72;isup-oli=0>;tag=gK0b66d829
> To: <sip:+14104159270 at 67.231.4.93>;tag=as0850c6db
> Call-ID: 285991942_79966325 at 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)