No Sound

[Guest fmonroy]

Hello guys, I've an issue here:

 

I setup testing environment on a computer: trial version of VG7 and trial version of HMP drivers.

 

I got 2 accounts freeworlddialup.com and configured the VG system to use one of them.

 

When VG's service is started I can see the 4 lines waiting for a call.

 

At other computer I'm using X-Lite to make the call, then I dial the VG system's SIP number and the call is stablished.

 

Using a "Get Numbers" module as start point of my script, then I send several tones and nothing is received.

 

Shutting down VG and using X-lite in the test system the calls work perfectly.

 

I wonder if some ports need to be forwarded from the router (using NAT) to the VG system.

 

Thanks in advance.

 

 

FM.

[SupportTeam]

DTMF tone detection is pretty poor when sending DTMF tones as inband over the internet links. You should ensure that the tones are sent only using "RFC2883". Easiest way to see if tones are sent as inband is to answer the call and start recording, and then press some tones from the callers phone. If you can hear the tones in the recorded sound file then the tones are sent as inband.

If you can hear the message that the IVR is playing then the ports are being forwarded OK.

 

If you are still having problems please post both the VoiceGuide traces and also make and post a WireShark network trace (see www.WireShark.org). This will let us see what information is being sent over the VoIP connection.

 

[Guest fmonroy]

Here are the traces. Please help me diagnose the problem. Trying to setup asterisk to have all the test system in the local network, do you have any guide to set it up?

 

Thanks in advance.

 

 

FM.

Traces.zip

[SupportTeam]

The supplied traces look fine, they show the sound file being played.

 

Have you tried calling the IVR system direct from another machine on the same LAN? Just try 'dialing' the IVR machines IP direct using a softphone on another machine or a VoIP handset on the same LAN. This should let you verify whether the sound is being played.

 

The WireShark trace did not contain any RTP packets, so it was not possible to see what sound was sent out. When posting WireShark traces in future please do not filter the posted trace.

[Guest fmonroy]

I'm using X-Lite to dial into the system. Turning VG off then starting X-Lite communication on each direction goes perfect.

 

Im' filtering the packets only on a host basis, no other packet drop, that can be the problem, the system does not send nor receive any sound.

 

Setting IVR host as DMZ does not help, so can be configuration problem. Is any configuration needed? I just only changed the SIP parameters and default TTS voice.

 

Do you have any guide to set up asterisk?

 

Thanks in advance.

 

 

FM.

[SupportTeam]

I'm using X-Lite to dial into the system. Turning VG off then starting X-Lite communication on each direction goes perfect.

OK, so the system is working when you dial directly into it.

 

I'm not sure what your network setup is like, but it sounds like you are trying to have calls come into the system over the public/general internet(?).

 

Here is one example of what ports need to be forwarded: http://www.voipconfig.com/port_forwarding.htm

 

Lookink at WireShark traces will let you see what RTP ports the negotiated by the VoIP connection as well.

 

We do not have guides on setting up Asterisk itself.