Hi guys,
I have searched the internet a lot but i cant find help for registering h323 user in asterisk using sjphone. All i can find is registring SIP or IAX users. So i will be very thankfull to the person who can help me.
ok, i’m just installing H323 on my Asterisk box at the office so i can see where you’re at … gimme a minute !
you are so nice…im waitin for ur helpfull reply.
Youi also have to install sjphone which u can download from www.sjlabs.com
i know, i know ! i never thought i would ever be looking at SJPhone again in my life !!
Well ! best of luck
incase u need any help for h323 installation, read the file
/asterisksource/channels/h323/README
ok, this confirms it. i don’t like SJPhone.
using netmeeting i can make and take calls no problem, just using the ip address of my asterisk box as a gateway. so the h323 implementation is working. there is no registration/gatekeeper process currently, although it seems that Jeremy is working on it.
do you need to use SJPhone ? have you tried using a gatekeeper ?
no i dont really need to use sjphone, i’ll use any phone which works with h323 protocol.
im using quintum as a gateway, not a gatekeeper.
i dont know how to use netmeeting. I mean i do have netmeeting installed on my pc but how to configure it with asterisk server?
also copy paste ur channel configuration for h323.conf file and also the context in extensions.conf file.
netmeeting is a doddle. under Options … Advanced Calling … Gateway Settings, tick the box, enter your Asterisk IP address and you’re done. then just enter your desired extension number as usual.
my h323.conf
[code][general]
port = 1720
bindaddr = 192.168.10.2
disallow=all
allow=gsm
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
gatekeeper = DISCOVER
AllowGKRouted = YES
context=from-internal
[tonyh323]
type=user
host=192.168.10.124
context=from-internal
incominglimit=1[/code]
and a snippet of extensions.conf
exten => 467,1,Dial(H323/192.168.10.124,30,r)
exten => 467,2,Hangup()
dial 467 from my SIP phone, netmeeting rings, dial 205 from netmeeting, my SIP phone rings. can’t be easier. i still don’t like SJPhone.
i’m leaving this for now, but i may look at a gatekeeper sometime later.
BAD NEWS
It doesnt work for me. Diaplays the following error:
– Executing Dial(“SIP/Ammad-fdc1”, “H323/192.168.0.77|20”) in new stack
– Called 192.168.0.77
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/Ammad-fdc1’ status is ‘CHANUNAVAIL’
Any Clue. Maybe h323 is not properly installed on my machine.
congrats 
finally it is working. i did some tweeking to my conf files, found some errors, and now its working fine. im making every kind of callls
H323 -> SIP
SIP ->H323
H323 ->H323
SIP ->SIP
thanx for ur help baconbutty, it made a lot of difference.
good news. did you ever tell us why you needed to use H323 ?
well i myself certainly did not want to use h323.
Actually im working for a telecom company as an internee. they are shifting from hardware pbx to software pbx. they have appointed me to test all protocols for asterisk. thats what im doing
what about u, what do u do?
self-employed IT consultant … means i seem to do anything that involves a computer, however remote !!
but i’m sat at home for a few weeks, getting over an op. not much fun, and the reason i’m posting so much here !!
hi can you help me
i am unable to establish rtp session between the phones
the call is getting established but
i cant really transmit voice
Please help
i used the config files in this mail but it cant be done
thanks in advance
bye
[quote=“bsasikiran”]hi can you help me
i am unable to establish rtp session between the phones
the call is getting established but
i cant really transmit voice
Please help
i used the config files in this mail but it cant be done
thanks in advance
bye[/quote]
like we can tell what the problem is without knowing anything about your setup !!
hi rizvan, i am doing project on voip in which i supposed to communicate between two h323 clients,please tell me what changes i should do in the h323.conf file and what softphone i should use
thanks