[Asterisk-Users] Calls connected, but no audio

Derek Whitten derek at kfuq.net
Mon May 29 12:13:18 MST 2006 

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
>>>>>>>>> _______________________________________________
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>>>>>>>>>
>>>>>>>>> Asterisk-Users mailing list
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>>>>>>>>>
>>>>>>>>
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>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
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you blocking the RTP ports?  (rtp.conf)



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