[Asterisk-Users] Calls connected, but no audio
Miles Scruggs
asterisk at garnetweb.com
Mon May 29 11:48:03 MST 2006
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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>>>>>>>
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>>>>>>
>>>>>
>>>>
>>>
>>
>
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