Nick Khamis wrote:
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: [email protected] <mailto:[email protected]>. Asterisk (Internal) <-> SIP Trunk (External) Call-ID: [email protected]:5060 <http://[email protected]:5060/>. SIP Trunk (External) "BYE" <-> OpenSIPS (Internal) Call-ID: [email protected]:5060 <http://[email protected]:5060/>. The call id was changed twice.... Could this be a two part problem?
Yes. Until you can isolate it more it's all just a guess but it still doesn't seem like a problem with Asterisk itself.
-- 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
