[size=150]hi, i have installed asterisk with default configuration for everything. i have also tested it with xlite and sjphone over LAN environment (Xlite -> Xlite(using SIP) and Sjphone->Xlite(using H323)).
[b]The problem im facing is that Sjphoen does not get registered on the asterisk server, so asterisk server does not no if there is any user over lan who wants to call. what i mean to say is, when i call from sjphone to xlite thru asterisk, asterisk recieves the call and routs the call to the user(xlite phone), who is registered, so it already knows that the called user exist.
But when i try to call another user using sjphone, it doesnt recognise them at all, as the user hasnt registered with asterisk using sjphone.
i have tried to use my own context in extensions.conf, and also i have made a channel using h323.conf file for sjphone but it calls the server directly and utilizez any s extension in any context.
here is my h323.conf channel for sjphone:
[Try1]
type=h323
host=dynamic
context=Riz
incominglimit=4
here is my context in the extensions.conf file:
[Riz]
exten=> 200,1,Dial(h323/Imtiaz)
exten=> 100,1,Dial(SIP/Ammad)
BUT it doesnt uses any of it. The asterisk shows the following error
Starting H323/ip$192.168.0.77:1083/4096 at default,100,1 failed so falling back to exten ‘s’
call gets automatically excepted in the Demo context which contains the s extension. and even if i cal 2 except for 100, the call uses the same context.
I have to use sjphone coz i want to test the server for h323 protocol.
u can also recommend another softphone which uses h323 and gets registered with asterisk before making/recieving calls.
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