[Asterisk-Users] Asterisk remains in the media path

[jimfl]

----- Original Message ----- 
From: "Jeremy McNamara"
To: <asterisk-users at lists.digium.com>
Sent: Monday, May 03, 2004 12:48 PM
Subject: Re: [Asterisk-Users] Asterisk remains in the media path


> brian wrote:
> 
> >Can't do it because you are changing from one technology to another.
> >
> >  
> >
> 
> Actually its cuz chan_h323 sucks like that.
> 
> 
> Jeremy McNamara

So does this mean you could get direct RTP steams between a SIP client and
a IAX2 client?  What about inband/out of band DTMF issues?

Thanks,
Jim

> 
> >>-----Original Message-----
> >>From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> >>admin at lists.digium.com] On Behalf Of Paul Berger
> >>Sent: Monday, May 03, 2004 10:29 AM
> >>To: Liste Asterisk
> >>Subject: [Asterisk-Users] Asterisk remains in the media path
> >>
> >>Hi all,
> >>Just a quick question: I have an H323 terminal and some MGCP phones
> >>connected to *, and when they call each other * remains in the media
> >>path no matter what (while I'd like to have the RTP stream directly
> >>between the phones).
> >>- mgcp.conf has canreinvite=yes
> >>- extension.conf doesn't contain any Dial() instance with t or T
> >>Did I forget something?
> >>Thanks,
> >>Paul