[Asterisk-Users] Calls connected, but no audio

[Miles Scruggs]

Well I just set the port to 5061, and no other devices on this end have 
that port.  I still have the same problems though.  The strange thing is 
that I have better luck calling the asterisk box itself rather than an 
outside line, but even that is intermittent.  Actually what I have found 
is that after my SIP device restarts I can call the asterisk box (but 
only once the second time it will not send audio), but I can't call an 
outside line, well it calls, answers, and bridges but no audio happens 
to pass.  I'm really confused.

Miles

Steve Totaro wrote:
> 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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