Re: [Asterisk-Users] Asterisk remains in the media path

2004-05-04 Thread Paul Berger
Le lun 03/05/2004 à 18:48, Jeremy McNamara a écrit :
> Actually its cuz chan_h323 sucks like that.

Correct me if I'm wrong, but I browsed the archives and I got the
feeling that you (Jeremy) were one of the main developers of the
chan_h323... aren't you a little harsh about your own work? :-)

Anyway, is there any plan in the chan_h323 roadmap to support direct RTP
between endpoints?

Thanks,
Paul

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Re: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread Eric Wieling
On Mon, 2004-05-03 at 12:05, jimfl wrote:
> 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?

IAX/IAX2 does not use RTP.

--Eric

-- 
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
"Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and
http://www.fnords.org/~eric/asterisk/ (my site) and
http://asteriskdocs.org/

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Re: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread Steven Critchfield
On Mon, 2004-05-03 at 12:05, jimfl wrote:

> 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?

IAX doesn't use rtp and therefore it couldn't do it either. All DTMF
should be OOB to be reliable.

-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread jimfl
- Original Message - 
From: "Jeremy McNamara"
To: <[EMAIL PROTECTED]>
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: [EMAIL PROTECTED] [mailto:asterisk-users-
> >>[EMAIL PROTECTED] 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

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Re: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread Jeremy McNamara
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







 

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] 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
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RE: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread Paul Berger
Le lun 03/05/2004 à 17:34, brian a écrit :
> Can't do it because you are changing from one technology to another.

Thanks for your answer.
H323 and MGCP are supposed to stay on the call control level, why isn't
it possible to open RTP channels between the terminals then?
Again, thanks,
Paul

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RE: [Asterisk-Users] Asterisk remains in the media path

2004-05-03 Thread brian
Can't do it because you are changing from one technology to another.

bkw

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] 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
>
>
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