[Freeswitch-users] No audio/dtmf from softphone behind NAT

Brian West brian at freeswitch.org
Mon Apr 5 10:34:09 PDT 2010


sofia loglevel all 0
sofia profile xx siptrace on

replace xx with profile.  What you have provided is NOT a sip trace.

Thanks,
Brian

On Apr 5, 2010, at 12:28 PM, Fraser Redmond wrote:

> Thanks Brian. Sorry, should have done a full sip trace before, but here is one now:
> 
> Calling an IVR dialplan:
> http://pastebin.freeswitch.org/12634
> 
> Calling from one extn to another.
> http://pastebin.freeswitch.org/12633
> (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.)
> 
> For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all.
> 
> Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.)
> 
> Cheers,
> Fraser

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