fangeli

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

Yes and not work.

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