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Sip=>H323 doesn´t work

jonhp3
Posts: 2
Member Since:
2008-07-22

Sip=>H323 doesn´t work

Hi Folks,

I´ve followed the guides existing on this forum and I´ve got the chanell ooh323 working on asterisk. I´m not using any gatekeeper or any other device and I want to make a call between two softphones on de same lan, one running SIP and the other running H323.
I´ve configured on the TB webui a custom estension with the dial string OOH323/"IP of the computer running the h323 softphone". My ooh323.conf file is like that:

[general]
port=1720
bindaddr="Trixbox server IP"
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gateway=no
gatekeeper = DISABLE
faststart=yes
h245tunneling=yes
context=default
rtptimeout=60
tos=lowdelay
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833

With this configuration I am able to make a h323=>SIP call dialing "SIPextNUMBER"@"TB server IP adress". But when I call the h323 ext using the sip phone the it rings but when I answer the call there is no audio transfer and the sip fone still rings! The call drops when I hangup the h323 phone proving that the signaling is partially working. I´ve no ideia what it must be. Please help me!



16again
Posts: 348
Member Since:
2007-03-04
try removing allow=gsm and

try removing allow=gsm and allow=alaw
afaik, ooh323 only supports ulaw and g729



jonhp3
Posts: 2
Member Since:
2008-07-22
Thanks for reading my

Thanks for reading my post!
I´ve tried changing the codecs and it has not worked too... The problem still remains! Any other ideas?



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