Sip to h323; h323 to sip


#1

hi everybody!

i would like to ask has anyone tried sip to h323 and h323 to sip calls?

using GNUGK and OOh323 driver module for asterisk 1.2

thank you very much.

ive tried but i cant make ti have a successful call.

DIAL PLAN:

[default]
exten => _X.,1,Dial(SIP/${EXTEN},60)
exten => _X.,2,Hangup
include => h323_in
include => to_h323

;from h323
[h323_in]
exten => _5.,1,Dial(SIP/&(EXTEN),60)
exten => _5.,2,Hangup

;to h323
[to_h323]
exten => _6.,1,Dial(H323/&(EXTEN),60)
exten => _6.,2,HangUp

OOH323.CONF

[general]
;Define the asetrisk server h323 endpoint
listenAddress=192.x.x.x
listenPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=10000

[test_2]
type=friend
context=h323_in
port=1720
h323id=sipman
ip=192.x.x.x
disallow=all
allow=ulaw
rtptimeout=60
dtmfmode=rfc2833

:smile:


#2

I have with an Avaya Prologix using ooh323. With v1.2.1 of Asterisk you need to address the H323 channel as follows:

Dial(OOH323/${EXTEN}@89.1.250.101|45);

If you still have problems, please post your CLI output when you attempt to dial.