ooh323 to sip / no ringback tone


Anyone have idea how to slove the problem ? I have configured Asterisk 1.2.6 and ooh323. It work fine in SIP, but it have some problem in h323 to sip. It have no ringback tone. Audio have no problem in both way after they bridge.

It may be the NAT problem I think. Any idea ? Please advice. Thanks a lot.

Best Regards,

I have a similar problem and I figured out that it may be related to audio stream channel not being established. I put the follow line at the beginning of the inbound call context.

And I added the “r” option to the Dial method

and it works. Please be aware that “r” option may cause problems.

Adding the “r” option will generate a ring tone to the caller even if it is not appropriate. But since my system will always answer the phone(voice mail or auto attendant), it is ok for me to just use ring tone.

Best Regards,