Patton 4100 fxo NO audio

[dexter]

3cx configured on 10.0.0.100:5060
firewall tests are ok

If I call from outside via pstn the inside extension rings but no audio on both sides
21:34:02.640 MediaServerReporting::RTPReceiver [MS105000] C:22.1: No RTP packets were received:remoteAddr=10.0.0.202:4872,extAddr=myexternalip:9000,localAddr=myexternalip:9000
21:34:02.640 Call::Terminate [CM503008]: Call(22): Call is terminated
21:34:02.640 Call::Terminate [CM503008]: Call(22): Call is terminated
21:34:02.625 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:10001 forwards to DN:103
21:34:02.625 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=anonymous
21:33:55.046 CallLeg:nConfirmed Session 913 of leg C:22.1 is confirmed
21:33:54.859 CallCtrl:nLegConnected [CM503007]: Call(22): Device joined: sip:[email protected]:5060;rinstance=df53620e1e00ee79
21:33:54.843 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:10001 forwards to DN:103
21:33:54.843 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=anonymous
21:33:54.843 CallCtrl:nLegConnected [CM503007]: Call(22): Device joined: sip:[email protected]:5062
21:33:54.843 MediaServerReporting::SetRemoteParty [MS210003] C:22.1:Answer provided. Connection(transcoding mode):94.36.68.162:9000(9001)
21:33:54.828 MediaServerReporting::SetRemoteParty [MS210001] C:22.2:Answer received. RTP connection: 10.0.0.23:42012(42013)
21:33:54.828 CallLeg::setRemoteSdp Remote SDP is set for legC:22.2
21:33:49.609 Extension:rintEndpointInfo [CM505001]: Ext.103: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [3CX Phone 6.0.727.0] Transport: [sip:10.0.0.100:5060]
21:33:49.609 CallCtrl:nAnsweredCall [CM503002]: Call(22): Alerting sip:[email protected]:5060;rinstance=df53620e1e00ee79
21:33:49.062 MediaServerReporting::SetRemoteParty [MS210002] C:22.2:Offer provided. Connection(transcoding mode): 10.0.0.100:7056(7057)
21:33:49.046 CallCtrl:nSelectRouteReq [CM503004]: Call(22): Calling: Ext:Ext.103@[Dev:sip:[email protected]:5060;rinstance=df53620e1e00ee79]
21:33:49.046 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]:5060]
21:33:49.046 MediaServerReporting::SetRemoteParty [MS210000] C:22.1:Offer received. RTP connection: 10.0.0.202:4872(4873)
21:33:49.046 CallLeg::setRemoteSdp Remote SDP is set for legC:22.1
21:33:49.046 Line:rintEndpointInfo [CM505002]: Gateway:[Patton Ciro Menotti] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA03F92F R5.2 2008-09-19 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26] Transport: [sip:10.0.0.100:5060]
21:33:49.046 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:10001 forwards to DN:103
21:33:49.031 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=anonymous
21:33:49.031 CallCtrl:nIncomingCall [CM503001]: Call(22): Incoming call from anonymous@(Ln.10001@Patton Ciro Menotti) to [sip:[email protected]:5060]
21:33:48.859 LineCfg::getInboundTarget [CM503012]: Inbound out-of-office hours' rule for LN:10001 forwards to DN:103
21:33:48.828 LineCfg::getInboundTarget Looking for inbound target: called=anonymous; caller=anonymous
21:33:48.781 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.202:5062;branch=z9hG4bKb07d293d7983cb3df
Max-Forwards: 70
Contact: [sip:[email protected]:5062]
To: [sip:[email protected]]
From: [sip:[email protected]:5062];tag=c272e66d89
Call-ID: 1d4df2fd831e63a9
CSeq: 7420 INVITE
Proxy-Authorization: Digest username="10001",realm="3CXPhoneSystem",nonce="12870390828:8eee1b2b51c2599dd3edc866a7cb19e1",uri="sip:[email protected]",response="65ed8fef27772a0b33676cafd64701b2",algorithm=MD5
Supported: replaces
User-Agent: Patton SN4112 JO EUI 00A0BA03F92F R5.2 2008-09-19 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26
Content-Length: 0



Also if I call from one internal extension to another everrything works, but if I try to call outside via pstn no way, here's the log
what does failed. Reason: Not Registered means?
gateway is configured not to require registration bot for incoming and outgoing calls

21:28:30.421 Call::Terminate [CM503008]: Call(21): Call is terminated
21:28:30.406 Call::RouteFailed [CM503015]: Call(21): Attempt to reach [sip:%[email protected]:5060] failed. Reason: Not Registered
21:28:30.406 Call::RouteFailed [CM503015]: Call(21): Attempt to reach [sip:%[email protected]:5060] failed. Reason: Not Registered
21:28:30.406 CallCtrl:nSelectRouteReq [CM503016]: Call(21): Target is not registered: Out.#053546824{rule=Asterisco}
21:28:30.359 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:%[email protected]:5060]
21:28:30.359 MediaServerReporting::SetRemoteParty [MS210000] C:21.1:Offer received. RTP connection: 10.0.0.23:42010(42011)
21:28:30.359 CallLeg::setRemoteSdp Remote SDP is set for legC:21.1
21:28:30.359 Extension:rintEndpointInfo [CM505001]: Ext.103: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [3CX Phone 6.0.727.0] Transport: [sip:10.0.0.100:5060]
21:28:30.343 CallCtrl:nIncomingCall [CM503001]: Call(21): Incoming call from Ext.103 to [sip:%[email protected]:5060]
21:28:30.343 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:%[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.23:5060;branch=z9hG4bK-d8754z-8b3cc64153007e17-1---d8754z-;rport=5060
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:%[email protected]:5060]
From: "3CXPhone"[sip:[email protected]:5060];tag=77401657
Call-ID: YWE4Y2Q4NGVkZjU0Nzg4MTIwZWY2MjM3MTkzMTEyYWE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Proxy-Authorization: Digest username="103",realm="3CXPhoneSystem",nonce="12870390510:5a35c114c929635528b874d384825f10",uri="sip:%[email protected]:5060",response="96c20b25ab12e06f6df4023db15e6073",algorithm=MD5
Supported: replaces
User-Agent: 3CX Phone 6.0.727.0
Content-Length: 0

I've tryed different settings, now extensions and gateway are confured to
-deliver pbx audio yes
-re-invite no
-replace header no

Iìm stucked, so please, tell me how to find a way out!
I thing this issue is with my patton confing...(made via standard 3cx config) I have 5.2version firmware