Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them:
----------------snip--------------------------------snip--------------------------------snip---------------- I am not certain of the reason for rejection but it has to do with the SDP, it does not seem to be a codec issue, the response is as you have seen: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.0.14;received=172.28.20.106;branch=z9hG4bK42d2ea03;rport=60017 From: "Andy" <sip:[email protected]>;tag=as5c784926 To: <sip:[email protected]>;tag=SD24jn898-4C46B8A2-5688CB2-0ADE2C09 Call-ID: [email protected] CSeq: 102 INVITE Reason: Q.850 ;cause=127 ;text="Interworking, unspecified" Content-Length: 0 There looks to be a non-standard element in your SDP that is not supported by any of the networks. ----------------snip--------------------------------snip--------------------------------snip---------------- Which configuration file is possibly incorrect in this scenario? What dumps are likely to be useful to me? Thanks, Andy -- _____________________________________________________________________ -- 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
