Thanks Kevin. Did it work with Asterisk 1.2 because it ignored it?
Why now? On Dec 20, 2010 3:28 PM, "Kevin P. Fleming" <[email protected]> wrote: > On 12/20/2010 11:46 AM, Dovey Forman wrote: >> 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>---> >> >> INVITEsip:[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) tox.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 > > This is pretty clear... your softphone is requiring support for a > private SIP extension called 'call-detail', and since Asterisk does not > support it, it cannot process the INVITE request. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: [email protected] > 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
-- _____________________________________________________________________ -- 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
