Problem with sjphone

[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:


here is my context in the extensions.conf file:

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$ 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.

its a simple problem guys…

noone has a clue::::WoW

[quote=“rizwan_hisham”]its a simple problem guys…

noone has a clue::::WoW[/quote]
i’ll be quite frank. imo, your use of bold and increased font size is rude. as is posting the same question in multiple threads. also, your lack of information is astounding.

and to top it all, if it’s that simple a problem you would have fixed it. but you haven’t. if you want to PM me with SSH details i’ll happily take a look.

Not to be nasty of anything…

…but I must admit, though my english is not the best, you made it x-times harder to read by using that font style and size. I actually had problems reading it. Realy.


The SJ Phone, Being a SIP Type Soft phone should follow the same
setup rules under sip.conf. Therefore, the use of “type=h323” should
not be neccesary. My SJ Phone is set up for IAX communication Via
extensions.conf andf iax.conf and it works fine. Perhaps giving
"type=friend" a try and unremark the max calls = 4 will work.
Hope this works for ya!

thanx for ur help kpm850. Could u also tell me how to setup the sjphone. and by the way im not using sjphone for sip communication, im using it for h323, so if u know setting up sjphone for h323, plz help

Are you needing help with the SJ Phone Software itself or Asterisk
also, in addition?

The reason I ask is if you are getting a register from your set up,
I might be more help from your approach than from my own?

Also, I am using Asterisk PoundKey 1.2.8. which for the SJ Phone
may or may not make a difference. (I don’t believe it will!) But let me
know what version you are using?

well, i think i need help for both. i need to setup sjphone to use the channel which i have configured in h323.conf file, just as xlite softphone uses sip.conf file to get registered with asterisk. The problem is sjphone does not get registered with asterisk so it doesnt make any call to that sjphone.

I am not using any Asterisk Poundkey. im using asterisk v1.2.8 with openh323 v1.17.1 and pwlib v1.9.0

problem solved guys !!!