<p>Thanks Kevin.</p>
<p>Did it work with Asterisk 1.2 because it ignored it?</p>
<p>Why now?</p>
<div class="gmail_quote">On Dec 20, 2010 3:28 PM, "Kevin P. Fleming" <<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>> wrote:<br type="attribution">> On 12/20/2010 11:46 AM, Dovey Forman wrote:<br>
>> Hi;<br>>><br>>> I am running asterisk 1.6 from Fonality (Trixbox PRO).<br>>><br>>> I am trying to initiate a call FROM a softphone client to asterisk<br>>> (either an internal 4 digit extension call) or an outside line via a SIP<br>
>> trunk.<br>>><br>>> In both cases, asterisk rejects the call with a 420.<br>>><br>>> In this case, it’s a call from x3992 to x4415<br>>><br>>> Does this require a change on the softphone for x-call-detail?<br>
>><br>>> <--- SIP read from UDP://x.x.x.x:5060 <<a href="http://10.247.1.126:5060">http://10.247.1.126:5060</a>>---><br>>><br>>> INVITEsip:4415@x.x.x.x:5060;transport=udp<br>>> <sip:4415@s144701.trixbox.fonality.com:5060;transport=udp>SIP/2.0<br>
>><br>>> To: <sip:4415@x.x.x.x5060;transport=udp<br>>> <sip:4415@s144701.trixbox.fonality.com:5060;transport=udp>><br>>><br>>> From: <sip:000000003992@x.x.x.x:5060<br>>> <<a href="http://sip:000000003992@10.247.1.126:5060">http://sip:000000003992@10.247.1.126:5060</a>>>;tag=4f5cb549<br>
>><br>>> Via: SIP/2.0/UDP<br>>> x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport<br>>><br>>> Call-ID: 350da2493d160e6f@ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ.<br>>><br>
>> CSeq: 1 INVITE<br>>><br>>> Contact: <sip:000000003992@x.x.x.x:5060<br>>> <<a href="http://sip:000000003992@10.247.1.126:5060">http://sip:000000003992@10.247.1.126:5060</a>>><br>>><br>
>> Max-Forwards: 70<br>>><br>>> Session-Expires: 1800<br>>><br>>> Min-SE: 90<br>>><br>>> Accept-Language: en<br>>><br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY<br>
>><br>>> Content-Type: application/sdp<br>>><br>>> *Require: x-call-detail*<br>>><br>>> Supported: timer<br>>><br>>> User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211<br>
>> (Windows NT 5.1)<br>>><br>>> Content-Length: 426<br>>><br>>> v=0<br>>><br>>> o=SIP 1292608808 1292608808 IN IP4 x.x.x.x<br>>><br>>> s=SIP<br>>><br>>> c=IN IP4 x.x.x.x<br>
>><br>>> t=1292608808 0<br>>><br>>> m=audio 10000 RTP/AVP 97 103 100 127 0 8 102 18 4 101<br>>><br>>> a=rtpmap:97 IPCMWB/16000<br>>><br>>> a=rtpmap:103 ISAC/16000<br>>><br>
>> a=rtpmap:100 EG711U/8000<br>>><br>>> a=rtpmap:127 EG711A/8000<br>>><br>>> a=rtpmap:0 PCMU/8000<br>>><br>>> a=rtpmap:8 PCMA/8000<br>>><br>>> a=rtpmap:102 iLBC/8000<br>
>><br>>> a=fmtp:102 mode=30<br>>><br>>> a=rtpmap:18 G729/8000<br>>><br>>> a=rtpmap:4 G723/8000<br>>><br>>> a=rtpmap:101 telephone-event/8000<br>>><br>>> <-------------><br>
>><br>>> --- (17 headers 17 lines) ---<br>>><br>>> == Using SIP RTP CoS mark 5<br>>><br>>> <--- Transmitting (no NAT) tox.x.x.x:5060 <<a href="http://10.247.1.126:5060">http://10.247.1.126:5060</a>>---><br>
>><br>>> SIP/2.0 420 Bad extension (unsupported)<br>>><br>>> Via: SIP/2.0/UDP<br>>> x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;received=x.x.x.x;rport=5060<br>>><br>>> From: <sip:000000003992@x.x.x.x:5060<br>
>> <<a href="http://sip:000000003992@10.247.1.126:5060">http://sip:000000003992@10.247.1.126:5060</a>>>;tag=4f5cb549<br>>><br>>> To: <sip:4415@x.x.x.x:5060;transport=udp<br>>> <sip:4415@s144701.trixbox.fonality.com:5060;transport=udp>>;tag=as34f3ff9f<br>
>><br>>> Call-ID: 350da2493d160e6f@ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ.<br>>><br>>> CSeq: 1 INVITE<br>>><br>>> User-Agent: Asterisk PBX 1.6.0.28<br>>><br>>> llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
>><br>>> Supported: replaces, timer<br>>><br>>> Date: Fri, 17 Dec 2010 18:00:04 GMT<br>>><br>>> *Unsupported: x-call-detail*<br>>><br>>> Content-Length: 0<br>> <br>> This is pretty clear... your softphone is requiring support for a <br>
> private SIP extension called 'call-detail', and since Asterisk does not <br>> support it, it cannot process the INVITE request.<br>> <br>> -- <br>> Kevin P. Fleming<br>> Digium, Inc. | Director of Software Technologies<br>
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>> skype: kpfleming | jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a><br>> Check us out at <a href="http://www.digium.com">www.digium.com</a> & <a href="http://www.asterisk.org">www.asterisk.org</a><br>
> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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