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Today's Topics:
1. Flash support (Sergio Garcia Murillo)
2. Re: How to find out if an incoming call has h324msupport
(Low Yu Siang)
3. got a bad video on minisip client (Mas Deva)
4. bad video quality on minisip client (Mas Deva)
5. Re: How to find out if an incoming call has h324msupport
(Klaus Darilion)
6. Re: How to find out if an incoming call has h324msupport
(Low Yu Siang)
7. Re: Flash support (Low Yu Siang)
8. Remotely disable video/audio stream in mobile phone (Low Yu Siang)
Message: 1
Date: Mon, 28 Jul 2008 20:52:56 +0200
From: Sergio Garcia Murillo <sergio.garcia@fontventa.com>
Subject: [Asterisk-video] Flash support
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <488E1588.6040103@fontventa.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi everyone,
As you may already know I'd been playing a bit with flash in the past.
I was thinking in finishing the development but first I would like to
pulse the interest on the following subjects:
-SWF playback.
This functionality would allow play a flash movie to an asterisk video
channel, mapping DMTF inputs to keys for user interaction.
-Flash video streaming
This functionality would allow to show an asterisk video channel on any
web page
-SIP-Flash gateway or asterisk flash channel
Best regards
Sergio
------------------------------
Message: 2
Date: Mon, 28 Jul 2008 19:59:37 -0700 (PDT)
From: Low Yu Siang <yusiang@yahoo.com>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <685158.2350.qm@web65601.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
Any idea how to find out if I am using chan_ss7? It doesn't carry transfer capability parameter and the patch 0010217 is only for chan_zap....
--- On Mon, 28/7/08, Klaus Darilion <klaus.mailinglists@pernau.at> wrote:
Quote:
From: Klaus Darilion <klaus.mailinglists@pernau.at>
Subject: Re: [Asterisk-video] How to find out if an incoming call has h324msupport
To: "Development discussion of video media support in Asterisk" <asterisk-video@lists.digium.com>
Date: Monday, 28 July, 2008, 3:20 PM
Joost Kuif | Mobillion schrieb:
> Hi Or,
>
> Use something like this:
>
> exten =>
s,10,GotoIf($[${TRANSFERCAPABILITY}!=DIGITAL]?11:20)
actually this checks only for digital calls, more accurate
is to check
the user information layer 1:
CHANNEL(userinformationlayer1)!=38
regards
kalus
>
> Grtz,
> Joost
>
>
------------------------------------------------------------------------
> *Van:* asterisk-video-bounces@lists.digium.com
> [mailto:asterisk-video-bounces@lists.digium.com]
*Namens *FastAgi FastAgi
> *Verzonden:* Sunday, July 27, 2008 1:00 PM
> *Aan:* asterisk-video@lists.digium.com
> *Onderwerp:* [Asterisk-video] How to find out if an
incoming call has
> h324msupport
>
> Hi,
> I installed h324m support for asterisk, and it works
great with phones
> supporting 3g video calls.
> I want to be able to distinguish phones that don't
support 3g video
> calls, send them to another context in the dialplan,
and make that a
> regular voice call.
> Is there any way to do that?
>
> Thanks,
> Or Agam
>
>
>
------------------------------------------------------------------------
>
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Message: 3
Date: Tue, 29 Jul 2008 11:01:10 +0700
From: "Mas Deva" <dvamust@gmail.com>
Subject: [Asterisk-video] got a bad video on minisip client
To: asterisk-video@lists.digium.com
Message-ID:
<5ab008890807282101h5646516fgdb86c5fb2b7a3671@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Now i'm in triying to build xcon in my lab.I think Asterisk server and video
mixer have already run normally.
But i got a problem when launch video. The result of the video that i
lounched with "videoswitch" command was so bad.(it was different from the
screnshoot on http://confiance.sourceforge.net/node/22).
here the display that i got from minisip log messages:
Message: 4
Date: Tue, 29 Jul 2008 11:07:41 +0700
From: "Mas Deva" <dvamust@gmail.com>
Subject: [Asterisk-video] bad video quality on minisip client
To: asterisk-video@lists.digium.com
Message-ID:
<5ab008890807282107n24f9d1eek7b9c4fec1df5aacf@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Dear all,
Now i'm in triying to build xcon in my lab.I think Asterisk server and video
mixer have already run normally.
But i got a problem when launch video. The result of the video that i
lounched with "videoswitch" command was so bad.(it was different from the
screnshoot on http://confiance.sourceforge.net/node/22).
here the display that i got from minisip log messages:
Message: 5
Date: Tue, 29 Jul 2008 10:16:36 +0200
From: Klaus Darilion <klaus.mailinglists@pernau.at>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <488ED1E4.505@pernau.at>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Low Yu Siang schrieb:
Quote:
Any idea how to find out if I am using chan_ss7? It doesn't carry
transfer capability parameter and the patch 0010217 is only for
chan_zap....
Hi! You are correct - reading the UL1 of incoming calls was not
implemented in chan_ss7. But it should be rather easy to do it yourself
- take a look at the patch for chan_zap and func_chan. The func_chan
part stays the same, you just have to port the chan_zap to chan_ss7 to
fill the channel structure with the UL1 information.
regards
klaus
------------------------------
Message: 6
Date: Tue, 29 Jul 2008 01:37:14 -0700 (PDT)
From: Low Yu Siang <yusiang@yahoo.com>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <49668.44927.qm@web65614.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
For time being, I am using an easier patch(but less accurate).
Will find some time to patch the UL1 later, thanks.
--- On Tue, 29/7/08, Klaus Darilion <klaus.mailinglists@pernau.at> wrote:
Quote:
From: Klaus Darilion <klaus.mailinglists@pernau.at>
Subject: Re: [Asterisk-video] How to find out if an incoming call has h324msupport
To: "Development discussion of video media support in Asterisk" <asterisk-video@lists.digium.com>
Date: Tuesday, 29 July, 2008, 4:16 PM
Low Yu Siang schrieb:
> Any idea how to find out if I am using chan_ss7? It
doesn't carry
> transfer capability parameter and the patch 0010217 is
only for
> chan_zap....
Hi! You are correct - reading the UL1 of incoming calls was
not
implemented in chan_ss7. But it should be rather easy to do
it yourself
- take a look at the patch for chan_zap and func_chan. The
func_chan
part stays the same, you just have to port the chan_zap to
chan_ss7 to
fill the channel structure with the UL1 information.
regards
klaus
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------------------------------
Message: 7
Date: Tue, 29 Jul 2008 04:45:51 -0700 (PDT)
From: Low Yu Siang <yusiang@yahoo.com>
Subject: Re: [Asterisk-video] Flash support
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <943028.80960.qm@web65605.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
SIP-Flash gateway sounds very interesting! A few weeks ago, I was trying to get app_swf running, unfortunately ended with gnash showing some weird output.
--- On Tue, 29/7/08, Sergio Garcia Murillo <sergio.garcia@fontventa.com> wrote:
Quote:
From: Sergio Garcia Murillo <sergio.garcia@fontventa.com>
Subject: [Asterisk-video] Flash support
To: "Development discussion of video media support in Asterisk" <asterisk-video@lists.digium.com>
Date: Tuesday, 29 July, 2008, 2:52 AM
Hi everyone,
As you may already know I'd been playing a bit with
flash in the past.
I was thinking in finishing the development but first I
would like to
pulse the interest on the following subjects:
-SWF playback.
This functionality would allow play a flash movie to an
asterisk video
channel, mapping DMTF inputs to keys for user interaction.
-Flash video streaming
This functionality would allow to show an asterisk video
channel on any
web page
-SIP-Flash gateway or asterisk flash channel
Best regards
Sergio
_______________________________________________
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http://www.api-digital.com--
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------------------------------
Message: 8
Date: Tue, 29 Jul 2008 04:53:35 -0700 (PDT)
From: Low Yu Siang <yusiang@yahoo.com>
Subject: [Asterisk-video] Remotely disable video/audio stream in
mobile phone
To: Development discussion of video media support in Asterisk
<asterisk-video@lists.digium.com>
Message-ID: <387438.88377.qm@web65607.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
Hi all,
I understand that this question is not related to asterisk....Can anyone tell me if it is possible to remotely disable transmission of video/audio stream in the mobile phone via h324m(h245) protocol?
Regards,
Low Yu Siang
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