RoadRunnR Posted July 31, 2015 Report Share Posted July 31, 2015 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted August 2, 2015 Report Share Posted August 2, 2015 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. Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.