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- Apr 2, 2008
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Hi all
Would need help with my configuration of 3CX PBX system. I have it behind two Routers, router1 is from my Internet provider, and has all the ports opened and redirected to my router2 (which belongs to me) and which is directly connected to the 3CX PBX. I have configured the router2 following the FAQ about the "Why does 3CX require static port mappings (Full Cone NAT)?" so I have opened ports 9000-9015 / port 5060 port 5090 and port 3478 as it is defined in the FAQ.
If I perform the firewall check, it passes and I can see in the results that ports 9000-9015 are ready for communications.
Then If I try to make outbound calls I can do them perfectly, no problem. But the problem is when I try to receive inbound calls. If I call my PBX from outside, I get the first tone, and then when somes picks it up, the call is disconected (cut).
Any idea of what I am missing???
Let me tell you something I can see in the Server Status.
STUN REGISTRATION.
I have these two stun servers in the Stun Server Option of my PBX
stun.terrasip.net --3478
stun.ekiga.net --3478
If I try to register my line using Stun servers, sometines it is registered, other times is not and I get this error message:
and when it is registered using STUN server if I try to call from outside, I dont even get a tone, the call get just cut.
So I have selected in the "Which IP to use in 'Contact' field for registration:" in the Registration options, the external IP address of Router1 (even it is dynamic).
If I select this option I get registered with my VOIP provider, I can get incoming calls, but when someones answers, the call is cut... and no comunication can be made.
This is what I get in the Server status after answering the call:
As soon as extension 001 answers, the people that made the call from outside gets the busy signal, or the call gets cut directly, and extension 001 does not hear anything, and sometimes in the logs I can see the "Non RTP message has been received message".
I was checking this system to use it as a Help Desk PBX tool, but I must say the working behaviour is not matching our expectatives, we dont stop getting situations like this one over and over, even when we think everything is configured, something like this happens.
Any ideas please?
Thanks a lot mates
Would need help with my configuration of 3CX PBX system. I have it behind two Routers, router1 is from my Internet provider, and has all the ports opened and redirected to my router2 (which belongs to me) and which is directly connected to the 3CX PBX. I have configured the router2 following the FAQ about the "Why does 3CX require static port mappings (Full Cone NAT)?" so I have opened ports 9000-9015 / port 5060 port 5090 and port 3478 as it is defined in the FAQ.
If I perform the firewall check, it passes and I can see in the results that ports 9000-9015 are ready for communications.
Then If I try to make outbound calls I can do them perfectly, no problem. But the problem is when I try to receive inbound calls. If I call my PBX from outside, I get the first tone, and then when somes picks it up, the call is disconected (cut).
Any idea of what I am missing???
Let me tell you something I can see in the Server Status.
STUN REGISTRATION.
I have these two stun servers in the Stun Server Option of my PBX
stun.terrasip.net --3478
stun.ekiga.net --3478
If I try to register my line using Stun servers, sometines it is registered, other times is not and I get this error message:
Code:
[CM506004]: STUN request to STUN server 194.39.182.241 has timed out; used Transport: 192.168.2.10:5060
and when it is registered using STUN server if I try to call from outside, I dont even get a tone, the call get just cut.
So I have selected in the "Which IP to use in 'Contact' field for registration:" in the Registration options, the external IP address of Router1 (even it is dynamic).
If I select this option I get registered with my VOIP provider, I can get incoming calls, but when someones answers, the call is cut... and no comunication can be made.
This is what I get in the Server status after answering the call:
Code:
20:03:25.093 Call::Terminate [CM503008]: Call(28): Call is terminated
20:03:25.093 Call::Terminate [CM503008]: Call(28): Call is terminated
20:03:25.078 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:03:25.078 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:03:25.062 Call::Terminate [CM503008]: Call(28): Call is terminated
20:03:25.062 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:03:25.062 InviteADS::onAckNotReceived [CM503018]: Call(28): ACK is not received
20:02:52.828 CallCtrl::onLegConnected [CM503007]: Call(28): Device joined: sip:[email protected]:5060;rinstance=2c458c12aa502374
20:02:52.828 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:02:52.812 CallCtrl::onLegConnected [CM503007]: Call(28): Device joined: sip:[email protected]:5060
20:02:52.812 MediaServerReporting::SetRemoteParty [MS210003] C:28.1:Answer provided. Connection(transcoding mode):213.60.109.15:9006(9007)
20:02:52.812 MediaServerReporting::SetRemoteParty [MS210001] C:28.2:Answer received. RTP connection: 192.168.2.21:42012(42013)
20:02:52.812 CallLeg::setRemoteSdp Remote SDP is set for legC:28.2
20:02:45.281 Extension::printEndpointInfo [CM505001]: Ext.001: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone v5.1] Transport: [sip:192.168.2.10:5060]
20:02:45.281 CallCtrl::onAnsweredCall [CM503002]: Call(28): Alerting sip:[email protected]:5060;rinstance=2c458c12aa502374
20:02:44.890 MediaServerReporting::SetRemoteParty [MS210002] C:28.2:Offer provided. Connection(transcoding mode): 192.168.2.10:7070(7071)
20:02:44.890 CallCtrl::onSelectRouteReq [CM503004]: Call(28): Calling: RingAll:800@[Dev:sip:[email protected]:5060;rinstance=2c458c12aa502374]
20:02:44.875 CallCtrl::onSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]:5060]
20:02:44.875 MediaServerReporting::SetRemoteParty [MS210000] C:28.1:Offer received. RTP connection: 86.109.97.3:32524(32525)
20:02:44.875 CallLeg::setRemoteSdp Remote SDP is set for legC:28.1
20:02:44.875 Line::printEndpointInfo [CM505003]: Provider:[TELSOME] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Asterisk PBX] Transport: [sip:192.168.2.10:5060]
20:02:44.875 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:02:44.859 CallCtrl::onIncomingCall [CM503001]: Call(28): Incoming call from 981585877@(Ln.10000@TELSOME) to [sip:[email protected]:5060]
20:02:44.687 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:10000 forwards to DN:800
20:02:44.656 CallLeg::onNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 86.109.97.3:5060;branch=z9hG4bK49e824d3;rport=5060
Contact: [sip:[email protected]]
To: [sip:[email protected]]
From: "981585877"[sip:[email protected]];tag=as3f45a961
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Date: Fri, 23 May 2008 18:02:51 GMT
User-Agent: Asterisk PBX
Content-Length: 0
As soon as extension 001 answers, the people that made the call from outside gets the busy signal, or the call gets cut directly, and extension 001 does not hear anything, and sometimes in the logs I can see the "Non RTP message has been received message".
I was checking this system to use it as a Help Desk PBX tool, but I must say the working behaviour is not matching our expectatives, we dont stop getting situations like this one over and over, even when we think everything is configured, something like this happens.
Any ideas please?
Thanks a lot mates