[asterisk-dev] RTP streams between H323 and SIP

[Johansson Olle E]

26 jun 2006 kl. 05.37 skrev Ken Chan:

> Hello,
> I am using Asterisk 1.2.9.1.
>
> I have a h323 phone (OpenPhone) and SIP Phone.  I make a
> call between those 2 phones.
>
> I set the "canreinvite=yes" for SIP phone.
>
> The RTP stream for the h323 phone still go through Asterisk.
> Anyone can give me a hand to fix this problem?
> I want the RTP streams to go from endpoint to endpoint
> without going through Asterisk.
>
> I have tried different h323 channel driver such as
> "chan_h323" (comes with Asterisk), Asterisk-Addon
> h323, asterisk-oh323 (from inAcess Network) and
> Objective Oh323 channel driver.
>
> None of them can solve my problem.  Any idea?

Well, it simply does not work that way, cross-channel.  Calls that go  
through more than one channel driver can't be
remotely bridged today.

/Olle


---
Olle E. Johansson * Asterisk Evangelist, developer * VOOP A/S
olle at voop.com