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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