Works in Version 3 but not Version 5 (NAT)

[waggy]

Hi
I have a system running fine in version 3, backed it up, and installed at a different location under version 5.1

If my phone (Snom 320) is running behind a natted router (external) then if I connect to V3 I can hear the other party. If I reconnect to the V5.1 then the other party can;t be heard. However - if I point the DMZ to the Snom 320 it works ok.

It works fine internally.

Any help very very gratefully received...

Logs attached

Version 5.1

12:53:44.000 Call::Terminate :[CM503008]: Call(47): Call is terminated
12:53:43.968 Call::Terminate :[CM503008]: Call(47): Call is terminated
12:53:34.984 CallCtrl:nLegConnected :[CM503007]: Call(47): Device joined: sip:[email protected]:5060
12:53:34.421 CallCtrl:nLegConnected :[CM503007]: Call(47): Device joined: sip:[email protected]:2059;line=gv8x1x75
12:53:30.671 Line:rintEndpointInfo :[CM505002]: Gateway:[sn4114] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.252:5060]
12:53:30.453 CallCtrl:nSelectRouteReq :[CM503004]: Call(47): Calling: VoIPline:10001@[Dev:sip:[email protected]:5060]
12:53:30.437 CallCtrl:nSelectRouteReq :[CM503004]: Call(47): Calling: PSTNline:10008@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]
12:53:30.390 CallCtrl:nIncomingCall :[CM503001]: Call(47): Incoming call from Ext.100 to [sip:[email protected];user=phone]
12:49:40.015 Call::Terminate :[CM503008]: Call(46): Call is terminated
12:49:39.984 Call::Terminate :[CM503008]: Call(46): Call is terminated
12:48:58.609 CallCtrl:nLegConnected :[CM503007]: Call(46): Device joined: sip:[email protected]:5060
12:48:58.593 CallCtrl:nLegConnected :[CM503007]: Call(46): Device joined: sip:[email protected]:2059;line=gv8x1x75
12:48:54.843 Line:rintEndpointInfo :[CM505002]: Gateway:[sn4114] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.252:5060]
12:48:54.703 CallCtrl:nSelectRouteReq :[CM503004]: Call(46): Calling: VoIPline:10001@[Dev:sip:[email protected]:5060]
12:48:54.671 CallCtrl:nSelectRouteReq :[CM503004]: Call(46): Calling: PSTNline:10008@[Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]
12:48:54.609 CallCtrl:nIncomingCall :[CM503001]: Call(46): Incoming call from Ext.100 to [sip:[email protected];user=phone]


Version 3
13:05:58.343 StratLink:nHangUp [CM104001] Call(337): Ext.100 hung up call; cause: BYE; from IP:80.229.0.16
13:05:43.125 CallLegImpl:nConnected [CM103001] Call(337): Created audio channel for Ln:10001@voipon (217.14.132.172:15936) with Media Server (217.207.4.181:9452)
13:05:43.093 StratInOut:nConnected [CM104005] Call(337): Setup completed for call from Ext.100 to Ln:10001@voipon
13:05:43.093 CallLegImpl:nConnected [CM103001] Call(337): Created audio channel for Ext.100 (80.229.0.16:64448) with Media Server (217.207.4.181:9450)
13:05:36.421 CallConf:nProvisional [CM103003] Call(337): Ln:10001@voipon is ringing
13:05:30.796 CallConf:nIncoming [CM103002] Call(337): Incoming call from 100 (Ext.100) to sip:[email protected]

 

[N_G]

Did you run the firewall checker and did it pass the test?

If not please review manual and ensure static port mappings....