Hi;
I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it’s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read from UDP://x.x.x.x:5060 <http://10.247.1.126:5060> ---> INVITE sip:[email protected]:5060;transport=udp<sip:[email protected]:5060;transport=udp> SIP/2.0 To: <sip:[email protected];transport=udp<sip:[email protected]:5060;transport=udp> > From: <sip:[email protected]:5060<http://sip:[email protected]:5060> >;tag=4f5cb549 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE Contact: <sip:[email protected]:5060<http://sip:[email protected]:5060> > Max-Forwards: 70 Session-Expires: 1800 Min-SE: 90 Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY Content-Type: application/sdp *Require: x-call-detail* Supported: timer User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211 (Windows NT 5.1) Content-Length: 426 v=0 o=SIP 1292608808 1292608808 IN IP4 x.x.x.x s=SIP c=IN IP4 x.x.x.x t=1292608808 0 m=audio 10000 RTP/AVP 97 103 100 127 0 8 102 18 4 101 a=rtpmap:97 IPCMWB/16000 a=rtpmap:103 ISAC/16000 a=rtpmap:100 EG711U/8000 a=rtpmap:127 EG711A/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (17 headers 17 lines) --- == Using SIP RTP CoS mark 5 <--- Transmitting (no NAT) to x.x.x.x:5060 <http://10.247.1.126:5060> ---> SIP/2.0 420 Bad extension (unsupported) Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;received=x.x.x.x;rport=5060 From: <sip:[email protected]:5060<http://sip:[email protected]:5060> >;tag=4f5cb549 To: <sip:[email protected]:5060;transport=udp<sip:[email protected]:5060;transport=udp> >;tag=as34f3ff9f Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq. CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.28 llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Date: Fri, 17 Dec 2010 18:00:04 GMT *Unsupported: x-call-detail* Content-Length: 0 --Dovey Forman
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