SIP softphone - no audio

I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
any audio behind NAT. Everything else seems to work, incoming & outgoing
calls reach SIP client & Asterisk server, but as soon as I should be hearing
something, nada.

I've tried changing SIP client description in Asterisk (nat=yes/no),
changing proxy type in X-Lite, nothing seems to help. What bothers me is
that incoming calls do get thru NAT (ringing notification), but audio does
not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
where SIP comms? What could be the problem here? Dumb-headed NAT box?

Outside NAT everything works just fine. I've googled the web & usenet
without results.

TIA,
Peter

Peter

You need to use nat = 1 instead of nat = yes

you sip.conf should be similar to this:

[general]
port=5060 ; port we use for sip - you can change this if you want
binadddr= sticktheipyouwantlisteningforsiphere
context = inbound ; this is where all the sip calls land
dtmfmode=inband ; to each his own
allow=ulaw

[Joe]
username=Joe
type=friend
secret=boink ; you dont need a secret if you havent set a password on the
sip device
host = puttheiphere ; or you can put dynamic if the sip devices ip changes
nat = 1 ; this is where you were in error.....
disallow=all ; just says we dont the codec set above
allow=ulaw ; this says ok we want ulaw
canreinvite = yes ; if this gives you problems change to no....


Testing with ulaw first is always best, some softphones only come with
ulaw....


-Greg




"Peter" <> wrote in message
news:bdo6ke$urdm1$...
> I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
> any audio behind NAT. Everything else seems to work, incoming & outgoing
> calls reach SIP client & Asterisk server, but as soon as I should be
hearing
> something, nada.
>
> I've tried changing SIP client description in Asterisk (nat=yes/no),
> changing proxy type in X-Lite, nothing seems to help. What bothers me is
> that incoming calls do get thru NAT (ringing notification), but audio does
> not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
> where SIP comms? What could be the problem here? Dumb-headed NAT box?
>
> Outside NAT everything works just fine. I've googled the web & usenet
> without results.
>
> TIA,
> Peter
>
>

last resort...

If you are still having problems , you need to forward a port on your router
to the ip of the sip device

so if your sip/softphone is 192.168.0.5

open up port 16384 and 5060 udp and send them to your phone...


16385,16386,16387,16388,16389 if you want the other functions......

-Greg


"Peter" <> wrote in message
news:bdo6ke$urdm1$...
> I'm trying out SIP softphones, X-Lite and SJPhone, and both does not have
> any audio behind NAT. Everything else seems to work, incoming & outgoing
> calls reach SIP client & Asterisk server, but as soon as I should be
hearing
> something, nada.
>
> I've tried changing SIP client description in Asterisk (nat=yes/no),
> changing proxy type in X-Lite, nothing seems to help. What bothers me is
> that incoming calls do get thru NAT (ringing notification), but audio does
> not. Shouldn't Asterisk be sending/receiving audio on the same UDP port as
> where SIP comms? What could be the problem here? Dumb-headed NAT box?
>
> Outside NAT everything works just fine. I've googled the web & usenet
> without results.
>
> TIA,
> Peter
>
>