[asterisk-users] SIP 420

[Kevin P. Fleming]

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@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@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@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: 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: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org