[Asterisk-Users] Calls connected, but no audio

[Steve Totaro]

Miles Scruggs wrote:
>
> Derek Whitten wrote:
>> Miles Scruggs wrote:
>>  
>>> Hmm all your questions are covered in this email, but I'll summarize it
>>> again in this reply:
>>>
>>> Server: 1.2.7.1 direct connection to the Internet
>>> config settings:
>>> [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
>>>
>>> Clients behind single NAT with a Linksys WRT54GS default settings
>>> Clients are 2 Eyebeam clients & 1 linksys PAP2-NA
>>>
>>> the audio has never worked consistently on the PAP2 only intermittently
>>> with better results in calling the asterisk box directly but only 
>>> rarely
>>> when calling outside lines.
>>>
>>> I have now set the phones to register every 60 seconds with no 
>>> change in
>>> results.
>>>
>>> There was no change in the 'sip show peers' as no settings were 
>>> changed,
>>> all you had requested was the output.
>>>
>>> finally the "yup everything is there" was in direct response to your
>>> statements in the previous email which asked me to confirm several 
>>> things.
>>>
>>> sip debug doesn't reveal anything more.
>>>
>>> I hope this summery helps
>>>
>>> Thanks
>>>
>>> Miles
>>>
>>>
>>> Steve Totaro wrote:
>>>    
>>>> N means NAT.  No N no NAT.
>>>>
>>>> Can you call now with audio in both directions?  Can you set the
>>>> phones to register every two minutes (expiration)?  Is the output from
>>>> sip show peers still the same before and after the audio working? 
>>>> Does sip debug give any info?  What type of router?
>>>> More info is good!  "yup everything is there" is a little hard to work
>>>> with.
>>>>
>>>> Is this a double NAT or is your asterisk box on a routable IP?  If it
>>>> is double NAT, forget it.
>>>> Thanks,
>>>> Steve
>>>>
>>>> Miles Scruggs wrote:
>>>>      
>>>>> 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
>>>>>>>>>>>
>>>>>>>>>>>                     
>>
>> you blocking the RTP ports?  (rtp.conf)
> This is just the default settings on the router, no ports are blocked.
>
> Miles

Not sure what else to tell you.  If the eyebeams work fine then the 
problem must be your in your linksys PAP2-NA.