Hi, I am testing asterisk in several situations, I still can not understand why we have to deal with two different h323 channels... Here is the problem, I have a cisco 3700 who sends h323 calls to asterisk. then I process the call upon several users parameters, and send it to another cisco gateway. All the transactions are made in h323. I first tried with h323 channel, and had no audio... I guess that is because g729 is not g729a. So I tried with oh323, and I finally got audio! Unfortunately the oh323 acts like a gatekeeper... and trys to create the direct call between the two ciscos... therefore, once the communication is accomplish it hangs up. I had to make available both g729a anf g729 in my oh323.conf, since the first cisco appears to use that codec. I tried all the combinations successfully, since cisco->asterisk->sip works ok and asterisk->cisco2 works too. I am sure this works since asterisk is transcoding can not make the rtp to flow directly between the two clients. Does somebody knows how to deal with this?
Thanks, Alito _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
