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:[email protected]:5060;transport=udp
<sip:[email protected]:5060;transport=udp>SIP/2.0
To: <sip:[email protected];transport=udp
<sip:[email protected]:5060;transport=udp>>
From: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>;tag=4f5cb549
Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport
Call-ID: 350da2493d160...@zhqtzgvsbdn2zhiwzjeuqu0usuruq09suc5orvq.
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060
<http://sip:[email protected]: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:[email protected]:5060
<http://sip:[email protected]:5060>>;tag=4f5cb549
To: <sip:[email protected]:5060;transport=udp
<sip:[email protected]:5060;transport=udp>>;tag=as34f3ff9f
Call-ID: 350da2493d160...@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: [email protected]
Check us out at www.digium.com & www.asterisk.org
--
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