[Asterisk-Users] Calls connected, but no audio

[Miles Scruggs]

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