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:4415 at x.x.x.x:5060;transport=udp<sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp> SIP/2.0 To: <sip:4415 at x.x.x.x5060;transport=udp<sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp>>From: <sip:000000003992 at x.x.x.x:5060<http://sip:000000003992 at 10.247.1.126:5060>>;tag=4f5cb549Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160e6f at ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ. CSeq: 1 INVITE Contact: <sip:000000003992 at x.x.x.x:5060<http://sip:000000003992 at 10.247.1.126:5060>>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:000000003992 at x.x.x.x:5060<http://sip:000000003992 at 10.247.1.126:5060>>;tag=4f5cb549To: <sip:4415 at x.x.x.x:5060;transport=udp<sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp>>;tag=as34f3ff9fCall-ID: 350da2493d160e6f at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101220/d14585cc/attachment-0001.htm>
On Mon, Dec 20, 2010 at 9:46 AM, Dovey Forman <dovey.forman at idt.net> wrote:> > 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?Yes. The softphone is requiring x-call-detail, which Asterisk does not support. The softphone either needs to drop that requirement completely, or change it to a Supported header so it can be processed by other SIP servers. -Jonathan
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:4415 at x.x.x.x:5060;transport=udp > <sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp>SIP/2.0 > > To: <sip:4415 at x.x.x.x5060;transport=udp > <sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp>> > > From: <sip:000000003992 at x.x.x.x:5060 > <http://sip:000000003992 at 10.247.1.126:5060>>;tag=4f5cb549 > > Via: SIP/2.0/UDP > x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport > > Call-ID: 350da2493d160e6f at ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ. > > CSeq: 1 INVITE > > Contact: <sip:000000003992 at x.x.x.x:5060 > <http://sip:000000003992 at 10.247.1.126:5060>> > > 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:000000003992 at x.x.x.x:5060 > <http://sip:000000003992 at 10.247.1.126:5060>>;tag=4f5cb549 > > To: <sip:4415 at x.x.x.x:5060;transport=udp > <sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp>>;tag=as34f3ff9f > > Call-ID: 350da2493d160e6f at 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: 0This 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: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org