[Asterisk-Users] Calls connected, but no audio

[Miles Scruggs]

The asterisk host is connected directly to the internet, the phones I am 
having issues with are behind NAT, but I'm only having issues with some 
of them.  Most specifically the phones on my linksys PAP2 adapter.  NAT 
at the remote location is provided via a standard out of the box config 
of a Linksys WRT54GS router.  Here are the settings for the PAP2:

[pap2]
type=friend
secret=something
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=private
callgroup=6
pickupgroup=6
callerid=name <1234567890>
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dtmfmode=rfc2833

This is a situation where I do have multiple SIP devices behind NAT, 
tell me more about using different port numbers for different devices, 
and what other things should I look out for?

Thanks

Miles


Steve Totaro wrote:
> You need to describe your NAT setup more.
> One thing to try is to set qualify to yes or a short number.  
> Essentially a keepalive for any routers in the middle.  If you have 
> multiple phones behind a remote NAT, make sure they are using 
> different ports.
>
> Miles Scruggs wrote:
>> Using sip connections some peers are not able to transmit or recieve 
>> audio.  All peers are setup the same aside from the NAT settings.  
>> The call will go through, called device will ring, but when it 
>> answers there is no audio connection.  From the callee, they will not 
>> here the rings, only silence when they dial the phone.
>>
>> The kicker is that sometimes it will work, and other times it will not.
>>
>> Miles
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