[Asterisk-Users] Calls connected, but no audio

[Miles Scruggs]

Steve Totaro wrote:
> 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.
Well I'm sure it does, but what I can't figure out is why it would work 
intermittently.  What is interesting is the eyebeams register on random 
ports such as:  24130, 8332, or 9240.  Does it even matter what port 
they try to connect on or is there some range that I can use?  is a STUN 
server going to help a situation like this?

Thanks

Miles