[asterisk-users] SIP Debugging Information..

[Matthew J. Roth]

Michael L. Young wrote:

> If I am reading this right, it looks like a BYE is coming in from
> the far end, Bandwidth.com.


Prior to that, Asterisk retransmits the OK to Bandwidth.com's INVITE
twice.  It doesn't look like Bandwidth.com receives any of them,
because they never respond with an ACK.  Since, from Bandwidth.com's
perspective, the call is never setup, they terminate it with a BYE.

It could just be a NAT issue, but there are two things I really don't
understand about the SIP dialog:

1) It starts with an ACK from Bandwidth.com.  Is it possible that
   the debugging output is missing the beginning of the dialog?

2) Every timestamp is "Nov 23 15:43:13".  I don't think the SIP
   session timers on either end should be expiring quickly enough
   for this to happen.

Do other calls originating from Bandwidth.com work properly?  If so,
comparing the SIP from a working call to a failed call may be
revealing.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer