[Asterisk-Users] asteisk, sip & NAT

[Hervé Thibaud]

Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit :
...
> I try to connect directly the both to fwd.pulver.com and now i have a
> perfect sound but the question is perhaps links after opening session 
> is only on the local networks with 10Mb/s.
> Once i can (when i'll have an external user to call) i'll try.

It's like i thought, the sound is nasty with many blanks when i try a
connection on internet and ISDN bandwith on one channel 64kb/s is not
enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and
use a bandwith of 64kb/s so. 
Is anybody that has a free or a very cheap solution (it's to try
asterisk) to have an IP phone hardware or software with G.723.1 codec

Other thing, I would like to try X-Lite but i have pb with registration,
i don't know how to write settings, i have an error (for example) : 
File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau
<sip:roseau at 192.168,0,1>" failed for '192,168,0,1'
I try many form of settings but didn't succeed.