Hi All I've had a system setup for a long time, but one issue has always been there and its come to a head.
I've always has problems with Asterisk not correctly selecting the call route for inbound DID's because the INVITE sent to it via my core (openSIPs) has the 'service number' not the DID in the INVITE. As per below INVITE sip:[email protected]:49640 SIP/2.0 Record-Route: <sip:202.11.11.11;lr=on;ftag=1ca96a43-co774-INS001;did=781.c6890334> Via: SIP/2.0/UDP 202.11.11.11;branch=z9hG4bK2ad4.58ca31d3.0 Via: SIP/2.0/UDP 202.13.13.13:5060;branch=z9hG4bK1ca53b72111cfdd3INV1ca96a43306 Max-Forwards: 34 Contact: <sip:[email protected]:5060> To: <sip:[email protected];user=phone> From: "0882229300"<sip:[email protected];user=phone;noa=national>;tag=1ca96a43-co774-INS001 Call-ID: 6f75-572-026197035530-IMG01-0-27.34.224.68 CSeq: 77401 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO Content-Type: application/sdp Date: Wed, 17 Oct 2012 03:09:42 GMT User-Agent: ENSR3.0.66.21-IS1-RMRG109-RG2100-CPO46 Content-Length: 309 On an inbound call from our wholesale supplier I run a command alias_db_lookup which I think changes the DID the service number. if (is_method("INVITE")) { ... ... ... if(alias_db_lookup("dbaliases","d")) { ... ... } I'm moved to supplying a commercially support Asterisk install for our customers and they (the commercial asterisk company) are saying that this rewrite is not correct and the INVITE should look like this. INVITE sip:[email protected]:49640 SIP/2.0 Any idea on how I might correct this ? What should the above INVITE really look like ? Thanks Mike
_______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
