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Cannot dial extensions or groups from outside

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mikedinatale

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Hi,

New isntall of the free edition as a trial for a Small Business system.

Using NexVortex tunking, right now with just one AASTRA 6757i CT phoen. All internal functions, in fact absolutely everytinhg including outbound calling works flawlessly. Incoming calls are answered by the digital receptionist... But when you dial a number, whether it be dial by name, an extension, etc, the call goes quiet and within 10 seconds disconnects.

Can anyone please help? Here is my server status log during the incoming call until disconnect.

09:08:37.522 Active calls counted toward license limit: [20]
09:08:34.169 [CM503008]: Call(19): Call is terminated
09:08:34.169 [CM503021]: Call(19): ACK is not received
09:08:24.495 [CM503008]: Call(18): Call is terminated
09:08:24.495 [CM503021]: Call(18): ACK is not received
09:08:23.444 [MS211000] C:20.1: 67.16.106.244:51182 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
09:08:15.471 [CM503007]: Call(20): Device joined: sip:[email protected]:40600;rinstance=239b0954278a7a5e
09:08:15.471 [CM503007]: Call(20): Device joined: sip:66.23.129.253:5060
09:08:15.455 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:[email protected]:5060]
09:08:15.455 [CM503002]: Call(20): Alerting sip:[email protected]:40600;rinstance=239b0954278a7a5e
09:08:15.316 [CM503025]: Call(20): Calling Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=239b0954278a7a5e]
09:08:15.285 [CM503004]: Call(20): Route 1: Ext:Ext.800@[Dev:sip:[email protected]:40600;rinstance=239b0954278a7a5e]
09:08:15.285 [CM503010]: Making route(s) to <sip:[email protected]:5060>
09:08:15.285 [CM505003]: Provider:[Nexvortex - US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:2165]
09:08:15.270 [CM503001]: Call(20): Incoming call from 7039090823@(Ln.10000@Nexvortex - US) to <sip:[email protected]:5060>
09:08:15.131 [CM503012]: Inbound out-of-office hours rule (unnamed) for 10000 forwards to DN:800
 
Do you see this in the logs?

09:08:34.169 [CM503021]: Call(19): ACK is not received

Go to Settings, network , stun server tab

Disable stun by checking Turn off stun server
Enter the public ip address in the box below and select the network card interface from the dropdown.

Let us know how this works for you.

(Normally this is not required but to be compatable with network topologies and routing scenarios out there, maybe this tip in your case will be beneficial)

make sure you have proper port forwarding for external rtp ports on the router to the pbx machine (default are 9000 - 9049 UDP)
 
I tried your suggestion and verified the firewall rules, but no change :( Any other thoughts? I really want to get this working so I can begin using it and selling the solution as a partner.

I have a pretty standard setup I believe. So when the message in the log "ack not received" who is not receiving the ack and who is supposed to be sending it? This might help me troubleshoot.

Mike
 
Ack is not received means that the ACK message in sip (the acknowledgement that confirms a session / call) was not received by the PBX.

Now this is an integral part of the sip protocol - without it sip will not work. So I am 100% sure that the provider is sending this - otherwise the provider will not work properly.

However sending it is not enough - the provider must send the ACK back to the correct ip address so I assume that the ack is being received but on a different IP / Port. \

This might occur because of routing issues or incorrect sip negotiation. You can make a wireshark capture and filter sip to see what is going on in reality.
 
I use NexVortex and they are configured to send to my external IP, I know because I had to configure it on their web admin site.

I'll check around in 3CX to make sure it isn't being sent improperly from there somewhere, but STUN is off now, and I have specified my external IP where it asks you to under VoIP providers > Advanced.

I don't know why this system is doign this though, I've used Response Point and Asterisk in the same network this weekend as tests, all using the same topology and addressing and those both worked fine.

Mike
 
The best thing to do like Nicky mentioned is to get a Wireshark of your 3CX machine. Also, Nexvortex should be able to provide you a trace of the call from their end for you to look at.

Compare the 2 and see where the ACK is being routed. There really isnt anything you can do to "fix" this until you know exactly what is happening.
 
Network topologies change, routing tables change. Caching changes constantly. Saying that asterisk and the other product worked is a bit irrelevant. The machine changed for sure.

I can tell you that I do not have this problem with my Nexvortex accounts. The Ack's always come to me exactly where I want them to come. But this again does not solve your problem just the same. Correct?

So base point - get a wireshark capture http://www.wireshark.org and you will have your answer in seconds.

BTW - What version of the pbx are you using?
 
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