HELP! How to set up two SIP communication?

Hi!

1.Enviroment:
	Server:192.168.8.1,runing  yate-service   (under dos command run "yate-console -v -l y.log"
	Client:192.168.8.163 ,using yate client,call "sip/sip:username@...:5060"
2.Problem:
 1) how to add SIP account? 
 2) server report error(see 3.1 below)

3.reference
3.1 server log
============================================================
Yate (676) is starting Thu Jun 01 11:19:16 2006
Loaded module Accounts from file
Loaded module Call Generator
Loaded module CdrBuild
Loaded module CdrFile
Loaded module Conference
Loaded module DirectSound
Loaded module MsgSniffer
Loaded module RegexRoute
Loaded module Registration from file
Loaded module RManager
Loaded module ToneGen
Loaded module WaveFile
Loaded module Wanpipe
Loaded module YRTP
Loaded module SIP Channel
Loaded module H.323 - based on OpenH323-SomethingOld
Loaded module IAX

trouble with h323chan

Excuse me for bad English.

I built Yate from tarball (0.9) and from CVS, and meet a problem: Yate
starts, but it doesn't even try to load h323chan.

From my yate.conf:

[general]
modload=enable
[modules]
h323chan.yate=enable

Output of run -vvvvvv:

Yate (15755) is starting Tue Jun 13 13:43:01 2006
<ALL> Engine::run()
<ALL> Plugin::Plugin() [0xb7f4c480]
Loaded module CdrBuild
<ALL> Plugin::Plugin() [0xb7f469f8]
Loaded module CdrFile
<ALL> Plugin::Plugin() [0xb7c95670]
Loaded module RegexRoute
<ALL> Plugin::Plugin() [0xb7c8ee84]
Loaded module Registration from file
<ALL> Plugin::Plugin() [0xb7c89ac0]
Loaded module Accounts from file
<ALL> Plugin::Plugin("tone") [0xb7c84a00]
Loaded module ToneGen
<tone:ALL> Building comfort noise at level -10

Re: trouble with h323chan

Thanks! The reason was that h323chan wasn't built...

Re: Routing with the register module

Re: New YAYPM 0.2 sources

Re: Routing with the register module

Thank you for your answer.

I agree and this is what I intend to do, but how do I designate this user 
concretely. For calling a user registered as "theuser" by SIP, what character 
string must I put as a value for the "callto" parameter of the "call.execute" 
or "call.route" message ?

When I call sip/sip:theuser <at> the_yate_machine, my impression is that regexroute 
takes precedence over the register module whatever priority value I put.

Le Mercredi 14 Juin 2006 09:07, vous avez écrit :
> my friend for registered user you dont have to write any thing in routing
> module.
>
> calls to registered user will automatically indentified and routed.
>
> Faisal.
>
> Etoile Dièse <support@...> wrote: Hello,
>
> what form may take an address to be routed by the register module ?
> - sip/(username registered) <at> localhost
> - sip/(username registered)
> - register/(username registered) <at> ???
>
> Does anybody know that ?
> Regards,

--

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Support Etoile Dièse