[Asterisk-Users] Calls connected, but no audio

[Steve Totaro]

SIP uses port 5060 by default.  Chances are your SIP phones are set to 
use port 5060 by default.  Some phones have a tick box that says "Use 
Random Port" or you can specify a port.  Start with port 5060 and move 
up so phone one would be 5060 phone two 5061 and so on.  The problem is 
most likely that your Linksys is mapping port 5060 to the phone that has 
last sent data which explains why it works sometimes but not others.  If 
your asterisk server is setup not to bind to a particular port for sip 
(sip.conf) then just try configuring the phones with unique ports and 
give it a try.

It is still a good idea to use qualify=yes in your asterisk (sip.conf) 
for each extension since it keeps port mappings open and active on your 
linksys.  Otherwise your Linksys port mapping may expire and an incoming 
call will be seen as unsolicited traffic and block it.

Thanks,
Steve Totaro

Miles Scruggs wrote:
> 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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