[Asterisk-Users] Calls connected, but no audio

[Miles Scruggs]

yup everything is there:

Name/username              Host            Dyn Nat ACL Port     Status   
pap2-2/pap2-2          123.123.123.123    D   N      5062     OK (93 ms)
pap2-1/pap2-1          123.123.123.123    D   N      5061     OK (39 ms)

I'm really confused why it has N for NAT when the sip settings listed in 
previous post have NAT set.

Thanks

Miles

Steve Totaro wrote:
> Make sure you have qualify=yes for each phone.  Type "sip show peers" 
> in the asterisk CLI and post the output when and when you are not able 
> to make calls.  Make sure that the new port settings are reflected in 
> asterisk.
>
> Miles Scruggs wrote:
>> 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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>>>>>
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>>>>
>>>
>>
>
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