[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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100405/2c891c8f/attachment.html
More information about the FreeSWITCH-users
mailing list