asterisk users - Nov 2006 - No sound: X-Lite -> Asterisk -> VoIP Provider -> Cellphone

Hi

I have the following setup to make outgoing calls:

X-Lite (build 34025) at home behind NAT -> Internet -> Asterisk at work 
behind NAT -> Internet -> VoIP provider -> GSM gateway -> cellphone.

I just tried calling my own cellphone, but there is no sound either way.

Here's what I did on the X-Lite at home in the Topology section:
IP address : Discover global address
STUN server : Discover server
Port used on local computer : Manually specify range 8000-8019

Here are the ports that I forwarded from my NAT router at home:
UDP 5060
UDP 3478 (STUN; needed?)
UDP 8000 to 8019

Is there something else I should do, either on my home setup or at work on 
the NAT router or Asterisk?

Thank you!

Hi Vincent -
> Here's what I did on the X-Lite at home in the Topology section:
> IP address : Discover global address
> STUN server : Discover server
> Port used on local computer : Manually specify range 8000-8019
>
> Here are the ports that I forwarded from my NAT router at home:
> UDP 5060
> UDP 3478 (STUN; needed?)
> UDP 8000 to 8019
>
> Is there something else I should do, either on my home setup or at work on
> the NAT router or Asterisk?
Just to double check - have you limited the RTP ports on the asterisk
server to 8000-8019 (in rtp.conf)?

Also, Xlite uses (or used to use) a silence suppresion mechanism that
doesn't work too well with asterisk.  According to the WIKI:

Turn off Silence Supression (to avoid RFC3389 warnings on Asterisk console):
Menu | Advanced System Settings | Audio Settings | Silence Settings |
Transmit Silence: Yes


- Noah

At 17:01 01/12/2006 +0100, "Noah Miller"
<noahisaacmiller@gmail.com> wrote:
>Just to double check - have you limited the RTP ports on the asterisk 
>server to 8000-8019 (in rtp.conf)?
Thanks. That what was missing. In rtp.conf, I fixed ports 10000-10019 and 
mapped those ports on the router, and it worked.

 >Also, Xlite uses (or used to use) a silence suppresion mechanism that 
doesn't work too well with asterisk.  According to the WIKI:
Turn off Silence Supression (to avoid RFC3389 warnings on Asterisk console):
Menu | Advanced System Settings | Audio Settings | Silence Settings |
Transmit Silence: Yes

OK. However, the person on the other end tells me that my voice was very 
low, barely audible. Do you know what could be done about it? Are there 
voice-related settings in Asterisk that I should look at?

Would playing with canreinvite to remove Asterisk from the loop and have 
RTP packets go directly from the VoIP provider to my X-Lite client at home 
make a difference? What should I do if canreinvite=yes means that the VoIP 
provider doesn't use the RTP ports that I expect to use on my side?

Thank you.