Nathan

ok i got the thing working now I changed the hunt group phone numbers in the AC gateway and that fixed it. Thanks

ok that was an easy fix thanks.

I set up a audiocodes MP118 fxo gateway for inbound and outbound calling the outbound works just fine but the inbound has issues. I know it has got to be an easy fix but the lack of an example really has me going in circles. the incoming calls are to ring ext.72 which is a group but the send call to extension box in the trunk configuration screen seems to do nothing. this is what a inbound call gives me [5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060: INVITE sip:172.16.20.30@172.16.20.30 SIP/2.0 Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558 Max-Forwards: 70 From: "201" <sip:201@172.16.20.32>;tag=1c206792810 To: <sip:172.16.20.30@172.16.20.30> Call-ID: 206792263163201111237@172.16.20.32 CSeq: 1 INVITE Contact: <sip:201@172.16.20.32:5060> Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005 Content-Type: application/sdp Content-Disposition: session Content-Length: 255 v=0 o=AudiocodesGW 206787991 206787869 IN IP4 172.16.20.32 s=Phone-Call c=IN IP4 172.16.20.32 t=0 0 m=audio 6050 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [5] 2011/03/16 11:50:52: Identify trunk (IP address/port and domain match) 2 [5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060: INVITE sip:172.16.20.30@172.16.20.30 SIP/2.0 Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558 Max-Forwards: 70 From: "201" <sip:201@172.16.20.32>;tag=1c206792810 To: <sip:172.16.20.30@172.16.20.30> Call-ID: 206792263163201111237@172.16.20.32 CSeq: 1 INVITE Contact: <sip:201@172.16.20.32:5060> Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005 Content-Type: application/sdp Content-Disposition: session Content-Length: 255 v=0 o=AudiocodesGW 206787991 206787869 IN IP4 172.16.20.32 s=Phone-Call c=IN IP4 172.16.20.32 t=0 0 m=audio 6050 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [5] 2011/03/16 11:50:52: SIP Tx udp:172.16.20.32:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558 From: "201" <sip:201@172.16.20.32>;tag=1c206792810 To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b Call-ID: 206792263163201111237@172.16.20.32 CSeq: 1 INVITE Content-Length: 0 [4] 2011/03/16 11:50:52: Call from account 201: Not an extension [5] 2011/03/16 11:50:52: Received incoming call without trunk information and user has not been found [5] 2011/03/16 11:50:52: SIP Tx udp:172.16.20.32:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558 From: "201" <sip:201@172.16.20.32>;tag=1c206792810 To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b Call-ID: 206792263163201111237@172.16.20.32 CSeq: 1 INVITE Contact: <sip:172.16.20.30@172.16.20.30:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/03/16 11:50:52: SIP Rx udp:172.16.20.32:5060: ACK sip:172.16.20.30@172.16.20.30 SIP/2.0 Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac206798558 Max-Forwards: 70 From: "201" <sip:201@172.16.20.32>;tag=1c206792810 To: <sip:172.16.20.30@172.16.20.30>;tag=e33de9132b Call-ID: 206792263163201111237@172.16.20.32 CSeq: 1 ACK Contact: <sip:201@172.16.20.32:5060> Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.6.20A.012.005 Content-Length: 0 I am not seeing how the AC gateway is sending the to information. does it just need to send to <sip:72@172.16.20.30>? thanks, Nathan

I set up the MP118 and out bound calls work just fine but in bound calls fail it says no route. I tried several different ways of setting up the incomming call routing from different toppics but none seem to be right. is there a tutorial out there about how to set this up so that a call goes from one of the pots lines to the snomone trunk number? Thanks ok now i have the snomone getting the call from the fxo to ip but it says call from account 201: not an extension. then after that Received incoming call without trunk information and user has not been found. then SIP Tx udp:172.16.20.32:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.16.20.32;branch=z9hG4bKac1680493678 From: <sip:201@172.16.20.32>;tag=1c1680489795 To: <sip:172.16.20.30@172.16.20.30>;tag=d46a52178e Call-ID: 16804892501532011173913@172.16.20.32 CSeq: 1 INVITE Contact: <sip:172.16.20.30@172.16.20.30:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 any help?