fangeli Posted December 20, 2013 Report Share Posted December 20, 2013 The problem is definitely an incorrect configuration. I created a sip trunk between an Asterisk and a snom one , the first have two trunks (one inbound and one outbound) type "SIP Gateway" pointing to a single trunk sull'asterisk . Calls from an extension registered on snom one, are rotated correctly toward the Asterisk, while the opposite does not happen. On the snom one I get the following error [8] 2013/12/20 12:33:26: Could not find a trunk (2 trunks) [7] 2013/12/20 12:33:26: Set packet length to 20 [6] 2013/12/20 12:33:26: Call-leg 24: Sending RTP for 5d667fb07f40a34e5884ff734539d1d8@192.168.XXX.4:5060 to 192.168.XXX.4:13694, codec not set yet [8] 2013/12/20 12:33:26: Incoming call: Request URI sip:601@192.168.YYY.4, To is <sip:601@192.168.YYY.4> [5] 2013/12/20 12:33:26: Received incoming call without trunk information and user has not been found [8] 2013/12/20 12:33:26: call port 24: state code from 0 to 404 [7] 2013/12/20 12:33:26: Set packet length to 20 [8] 2013/12/20 12:33:26: Hangup: Call 24 not found [8] 2013/12/20 12:33:26: Clearing call port 24, SIP call id 5d667fb07f40a34e5884ff734539d1d8@192.168.XXX.4:5060 I have checked all of the trunk and outbound route of the Asterisk, without finding obvious problems. Thank for the help Quote Link to comment Share on other sites More sharing options...
Vodia support Posted December 20, 2013 Report Share Posted December 20, 2013 Have you tried adding the Asterisk IP address in the trunk setting "Explicitly list addresses for inbound traffic" on the Vodia PBX? Quote Link to comment Share on other sites More sharing options...
fangeli Posted December 20, 2013 Author Report Share Posted December 20, 2013 Yes and not work. Quote Link to comment Share on other sites More sharing options...
Vodia support Posted December 20, 2013 Report Share Posted December 20, 2013 Can you post the full SIP trace? Thx Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 21, 2013 Report Share Posted December 21, 2013 The PBX needs to find out where the call comes from. It does that by going through a couple of rules (see http://wiki.snomone.com/index.php?title=Inbound_Calls). When the PBX says "Received incoming call without trunk information and user has not been found" it was not able to match the SIP request to any of the provided IP addresses in any trunk. I guess you need to double check the source IP of the request coming from the Asterisk server. Quote Link to comment Share on other sites More sharing options...
fangeli Posted December 23, 2013 Author Report Share Posted December 23, 2013 Thanks to all for the answers, now I understand where is the problem. I need to figure out how to use the "Send call to extension" option to route the incoming calls to various extensions. Currently if I call the extension 601, 602 or 603 from the main pbx, the call stops within the snom one with the error mentioned in the first post. I have read the document on the management of inbound call, but I have not figured out how to do this configuration. Thanks to all Quote Link to comment Share on other sites More sharing options...
Henrikt Posted January 6, 2014 Report Share Posted January 6, 2014 I use this setup in "Send call to extension": !([0-9]*)!\1!t!yy! where yy are the extension the call route to if nothing else match in the number plan. Best Henrikt Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 6, 2014 Report Share Posted January 6, 2014 I use this setup in "Send call to extension": !([0-9]*)!\1!t!yy! where yy are the extension the call route to if nothing else match in the number plan. In V5 we have turned that into a small JavaScript that offers some popular patterns. Whenever it comes to ERE, the road becomes slippery! Quote Link to comment Share on other sites More sharing options...
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