[asterisk-users] SIP 420

Dovey Forman
Mon, 20 Dec 2010 17:46:42 -0000

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=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) 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=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





--Dovey Forman
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