Nick Khamis wrote:
Hey Joshua, It was a poor choice of words on my part. What I meant to say was whether the problem was due to our asterisk configuration re-writing the RR when initiating the INVITE to our SIP trunk provider. Not sure if you had looked at the SIP trace included in the original email? If not I can resend it.
I saw, but my response stands. Asterisk does not rewrite anything. The outgoing leg to your SIP trunk is completely separate, it is not a forwarded/modified INVITE. With the information you have available I don't think Asterisk is the problem here. The traces also illustrate this, the BYE in the trace is from a completely different call than the other messages. (You can see by looking at the Call-ID).
-- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
