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
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