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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