[asterisk-users] SIP 420

[Dovey Forman]

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" <kpfleming at digium.com> 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: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
>
> --
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