On Tue, Apr 9, 2013 at 3:04 PM, Joshua Colp <jc...@digium.com> wrote:
> 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<http://lists.digium.com/mailman/listinfo/asterisk-users> > Hello Joshua, Thanks again for your response. I can understand how * does not rewrite anything. When you mention the difference in call id, are you referring to: UA <-> OpenSIPS <-> Asterisk (Internal) Call-ID: 595ad334-f06e97fa-3bbc8137@192.168.2.11. Asterisk (Internal) <-> SIP Trunk (External) Call-ID: 5a5fb47111cadd6146746c4446a1790c@70.10.163.44:5060. SIP Trunk (External) "BYE" <-> OpenSIPS (Internal) Call-ID: 705605f129adbf5a38b5a0ff72de8f39@70.10.163.44:5060. The call id was changed twice.... Could this be a two part problem? N.
-- _____________________________________________________________________ -- 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