snombtw Posted February 3, 2011 Report Share Posted February 3, 2011 Hi, We are meeting a problem with Snom-Pbx and 4-5 Snom phones : All incoming calls arrive on the same phone, direct calls all arrive on the same phone. We want to receive direct calls on the IP-phones, but all calls arrive always on the same phone. Here is the log found on the SnomOnePBX server : " [9] 2011/02/03 15:10:29: SIP Rx udp:91.121.X.X:5060: INVITE sip:003336672XXXX@192.168.X.X:5060;transport=udp;line=c81e728d SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net Contact: <sip:91.121.X.X:5060> Content-Type: application/sdp CSeq: 791574727 INVITE From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583 Max-Forwards: 30 To: <sip:0320xxxxxx@91.121.X.X;user=phone> User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8 Content-Length: 481 v=0 o=cp10 129674225667 129674225667 IN IP4 10.7.X.X s=SIP Call c=IN IP4 91.121.X.X t=0 0 m=audio 33172 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 [9] 2011/02/03 15:10:29: UDP: Opening socket on 0.0.0.0:60170 [9] 2011/02/03 15:10:29: UDP: Opening socket on 0.0.0.0:60171 [9] 2011/02/03 15:10:29: UDP: Opening socket on [::]:60170 [9] 2011/02/03 15:10:29: UDP: Opening socket on [::]:60171 [5] 2011/02/03 15:10:29: Identify trunk (line match) 2 [9] 2011/02/03 15:10:29: Resolve 667271: aaaa udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667271: a udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667271: udp 91.121.129.17 5060 [9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8 From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583 To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net CSeq: 791574727 INVITE Content-Length: 0 [7] 2011/02/03 15:10:29: Set packet length to 30 [6] 2011/02/03 15:10:29: Sending RTP for 10360-NL-2f3ab14a-2818812b3@sip.ovh.net to 91.121.X.X:33172, codec not set yet [8] 2011/02/03 15:10:29: Call from an trunk 2 [8] 2011/02/03 15:10:29: Trunk OVH Sip@snompbx.mydomain.fr has country code 33, area code not set [9] 2011/02/03 15:10:29: Incoming: formatted From is = "0678XXXXXX" <sip:+33678xxxxxx@sip.ovh.net;user=phone> [9] 2011/02/03 15:10:29: Incoming: formatted To is = <sip:+3332xxxxx@91.121.X.X;user=phone> [8] 2011/02/03 15:10:29: To is <sip:+33320xxxxxx@91.121.X.X;user=phone>, user 2, domain 1 [8] 2011/02/03 15:10:29: To user 41 [5] 2011/02/03 15:10:29: Domain trunk OVH Sip@snompbx.mydomain.fr could not identify user for +33366XXXXXX [7] 2011/02/03 15:10:29: Set packet length to 30 [9] 2011/02/03 15:10:29: Resolve 667272: aaaa udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667272: a udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667272: udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8 From: "0678XXXXXX" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583 To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net CSeq: 791574727 INVITE Contact: <sip:0033366xxxxxx@192.168.X.X:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3950 Content-Length: 0 [9] 2011/02/03 15:10:29: Resolve 667273: aaaa udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667273: a udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: Resolve 667273: udp 91.121.X.X 5060 [9] 2011/02/03 15:10:29: SIP Tx udp:91.121.X.X:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8 From: "0678xxxxxx" <sip:0678xxxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583 To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net CSeq: 791574727 INVITE Contact: <sip:0033366xxxxxx@192.168.X.X:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3950 Content-Length: 0 [9] 2011/02/03 15:10:29: SIP Rx udp:91.121.X.X:5060: ACK sip:0033366xxxxxx@192.168.X.X:5060;transport=udp;line=c81e728d SIP/2.0 Call-ID: 10360-NL-2f3ab14a-2818812b3@sip.ovh.net Contact: <sip:91.121.1X.X:5060> CSeq: 791574727 ACK From: "0678xxxxxx" <sip:0678xxxxxx@sip.ovh.net;user=phone>;tag=10360-MU-2f3ab14b-02c616583 Max-Forwards: 30 To: <sip:0320xxxxxx@91.121.X.X;user=phone>;tag=f86e8c263c User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.X.X:5060;branch=z9hG4bK-66B-160C7E8 Content-Length: 0 [8] 2011/02/03 15:10:29: Hangup: Call 444 not found " Some help will be very useful, THX Quote Link to comment Share on other sites More sharing options...
Porter Posted February 3, 2011 Report Share Posted February 3, 2011 Do you have the DIDs configured on each extension account? The DIDs should be in the "Account number(s)" field, (after the extension number) not in the ANI. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 3, 2011 Report Share Posted February 3, 2011 Right, you can enter the phone number like this "41 003336612345 00336643321" ("Account number(s)", space between the numbers). This means that the account has the (primary) name 41, and alias names 003336612345 and 00336643321. Then incoming calls can be matched against these names. Quote Link to comment Share on other sites More sharing options...
snombtw Posted February 3, 2011 Author Report Share Posted February 3, 2011 Hi, Thanks for the quick answer. Yes i have the DID (with ou without the country code) configured after the internal extension (e.g 41 32047471) When i set the "send call to extension 41" in the trunk parameters, calls are corectly routed on the end-user phone. As far as i can understand the logs, the DID is recognized as the log gives the correct extension number [8] 2011/02/03 15:10:29: To is <sip:+33320xxxxxx@91.121.X.X;user=phone>, user 2, domain 1 [8] 2011/02/03 15:10:29: To user 41 THis call was for user with extension 41. It just looks like after, instead of ringing the extension 41, it tries to forward the call to the SIP trunk? Thanks again Quote Link to comment Share on other sites More sharing options...
snombtw Posted February 6, 2011 Author Report Share Posted February 6, 2011 Hello Everybody, Has anyone any clue, idea, to help resolving this case ? Thanks a lot Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 7, 2011 Report Share Posted February 7, 2011 Yes i have the DID (with ou without the country code) configured after the internal extension (e.g 41 32047471) When i set the "send call to extension 41" in the trunk parameters, calls are corectly routed on the end-user phone. The 32047471 is probably not correct. Either enter 032047471 or to make it 100 % clear +3332047471. Quote Link to comment Share on other sites More sharing options...
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