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Extension always gets 404 Forbidden


RoadRunnR
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I have a SIP FXS (Telefon/FAX machine to SIP) gateway that refuses to play with a SNOM ONE.

 

Incoming calls work, but outgoing calls get rejected by the SNOM ONE with 403 Forbidden.

 

I already tried to configure it a SIP Trunk/Gateway, but then calls from the FXS box, through the SNOM OUT to our SIP trunk get rejected with 404 Not found. This seems to be a case of Trunk-To-Trunk routing not working, but I can't find any information on how to route between SIP trunks.

 

Any hints?

 

Here is a trace with the extension only config:

[5] 17:35:14.575	PACK:	SIP Rx udp:192.168.13.181:5060:
INVITE sip:6255555@192.168.13.8 SIP/2.0
From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead
To: <sip:6255555@192.168.13.8>
Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2
Max-Forwards: 70
Contact: <sip:6255555@192.168.13.181:5060>
Content-Type: application/SDP
Content-Length:171

v=0
o=SIP 25021527 1492432404 IN IP4 192.168.13.181
s=-
c=IN IP4 192.168.13.181
t=0 0
a=ptime:30
m=audio 2076 RTP/AVP 4 18 0 101
a=rtpmap:101 telephone-event/8000

[5] 17:35:14.576	PACK:	SIP Tx udp:192.168.13.181:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2
From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead
To: <sip:6255555@192.168.13.8>;tag=034a7cf432
Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8
CSeq: 1 INVITE
Content-Length: 0


[5] 17:35:14.577	PACK:	SIP Tx udp:192.168.13.181:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2
From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead
To: <sip:6255555@192.168.13.8>;tag=034a7cf432
Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8
CSeq: 1 INVITE
User-Agent: Vodia-PBX/5.2.6
Content-Length: 0


[5] 17:35:15.077	PACK:	SIP Tr udp:192.168.13.181:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.13.181:5060;branch=z9hG4bK-3a4fd3be-2bed76-35e2
From: <sip:sip@192.168.13.8>;tag=c0a80db5-13c4-3a4fd3be-2bed72-5ead
To: <sip:6255555@192.168.13.8>;tag=034a7cf432
Call-ID: 78f738-c0a80db5-13c4-3a4fd3be-2bed6e-ad7@192.168.13.8
CSeq: 1 INVITE
User-Agent: Vodia-PBX/5.2.6
Content-Length: 0

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The PBX needs to know which extension needs to be "charged" for the call. It does so by looking at the From header. There it finds the extension "sip". If that extension does not exist, the call will fail. You probably have to put the extension number some where into the device.

 

If you really wanted to use the extension number "sip" please make sure that you have a dial plan in the domain and that the extension uses it.

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