[Asterisk-video] asterisk-video Digest, Vol 18, Issue 10
Walter Rodrigues Filho
walter at telebit.net.br
Mon Oct 15 20:51:04 CDT 2007
MESSAGE 1:
Hi Emmanuel,
Regarding your request...
> On the other hand, if we need to proceed with videocaps back port, would
> someone be interested in testing video interoperability with us? We have
> here some various videophone, hard, soft, Wifi, etc. We could also
> provide the row source tree of our patched version.
We would be pleased to help.
We have got here also a variety of videophones like GXV 3000, Bria, Eyebeam,
etc.
We also have lab machines able to be accessed remotely to do this sort of
things.
You can reach us by email: walter at telebit.net.br
We are located in Belo Horizonte, Brazil GMT -3.
Waiting for your contact...
Regards,
Walter Rodrigues Filho
Telebit Telecom
MESSAGE 2:
Hi Klaus!
We met at OPENSER Meeting earlier this year in Paris.
Could you eventually help me compile mpeg4IP? The release I have been trying
to use seems to be corrupted.
Thanks for any hint.
Regards,
Walter R Filho
Telebit Telecom
walter at telebit.net.br
Em 11.10.07 14:00, "asterisk-video-request at lists.digium.com"
<asterisk-video-request at lists.digium.com> escreveu:
> Today's Topics:
>
> 1. videocaps - the return (Emmanuel BUU)
> 2. Re: app_h324m crash (Reza Fatahillah)
> 3. Re: app_h324m crash (Klaus Darilion)
> 4. Re: app_h324m crash (Reza Fatahillah)
> 5. Re: Pach for q931 in misdn (Rene van Weert)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 10 Oct 2007 23:18:23 +0200
> From: Emmanuel BUU <emmanuel.buu at ives.fr>
> Subject: [Asterisk-video] videocaps - the return
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <470D419F.1000801 at ives.fr>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hello everybody,
>
> We have been working a lot on video codec negociation on Astersik 1.4.x
> (http://bugs.digium.com/view.php?id=9815). The patch that corrects the
> various cases has become so huge that I have no hope it will be ever
> approved as a fix in the 1.4 branch. Except for some identified cases
> (183 Call progess with video playback), we reached a point where we need
> to back port videocaps into 1.4 branch to make any progress on video
> interoperability between sip phones. Before proceeding, we would like to
> ask again an update on videocaps or equivalent on the trunk.
>
> Is there any news on that matter? What are the plans?
>
> On the other hand, if we need to proceed with videocaps back port, would
> someone be interested in testing video interoperability with us? We have
> here some various videophone, hard, soft, Wifi, etc. We could also
> provide the row source tree of our patched version.
>
> Emmanuel BUU
> IV?S
> http://www.ives.fr/
>
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 10 Oct 2007 21:02:14 -0700 (PDT)
> From: Reza Fatahillah <ezhot_95 at yahoo.com>
> Subject: Re: [Asterisk-video] app_h324m crash
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <85303.52469.qm at web50403.mail.re2.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
>
> --- Sergio Garcia <sergio.garcia at fontventa.com> wrote:
>
>>
>> I've finally commited the patch to solve Reza's
>> crashes.
>>
>> Testing is welcome.. :)
>>
>> BR
>> Sergio
>
> Hi Sergio,
>
> Thanks for the patch.
> I have tried the callout several times. Still no
> crash.
> I think it's ok now.
>
> But, another problem is sometime the callout
> disconnected, with this message:
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
> (0) Spare: 0 Location: Transit network (3)
> Ext: 1 Cause: Temporary failure (41), class =
> Network Congestion (resource unavailable) (2) ]
>
> This happened also from previous version. The first
> call succeeded, then second call failed.
>
> Regards,
> Reza
>
>
>
> ______________________________________________________________________________
> ______
> Don't let your dream ride pass you by. Make it a reality with Yahoo! Autos.
> http://autos.yahoo.com/index.html
>
>
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 11 Oct 2007 08:14:45 +0200
> From: Klaus Darilion <klaus.mailinglists at pernau.at>
> Subject: Re: [Asterisk-video] app_h324m crash
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <470DBF55.3000308 at pernau.at>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
>
> Reza Fatahillah schrieb:
>>
>> But, another problem is sometime the callout
>> disconnected, with this message:
>> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard
>> (0) Spare: 0 Location: Transit network (3)
>> Ext: 1 Cause: Temporary failure (41), class =
>> Network Congestion (resource unavailable) (2) ]
>>
>> This happened also from previous version. The first
>> call succeeded, then second call failed.
>
> Does this happen during call setup or during the call? I often see call
> setup problems which I guess are located somewhere in the mobile network.
>
> regards
> klaus
>
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 11 Oct 2007 00:38:22 -0700 (PDT)
> From: Reza Fatahillah <ezhot_95 at yahoo.com>
> Subject: Re: [Asterisk-video] app_h324m crash
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <108764.40292.qm at web50409.mail.re2.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> This happened during call setup, after outgoing call
> preceeding. Then went to Disconnect Indication state.
>
> Reza
> --- Klaus Darilion <klaus.mailinglists at pernau.at>
> wrote:
>> Reza Fatahillah schrieb:
>>>
>>> But, another problem is sometime the callout
>>> disconnected, with this message:
>>> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU)
>> standard
>>> (0) Spare: 0 Location: Transit network (3)
>>> Ext: 1 Cause: Temporary failure (41), class =
>>> Network Congestion (resource unavailable) (2) ]
>>>
>>> This happened also from previous version. The
>> first
>>> call succeeded, then second call failed.
>>
>> Does this happen during call setup or during the
>> call? I often see call
>> setup problems which I guess are located somewhere
>> in the mobile network.
>>
>> regards
>> klaus
>
>
>
>
> ______________________________________________________________________________
> ______
> Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings,
> and more!
> http://tv.yahoo.com/collections/3658
>
>
>
> ------------------------------
>
> Message: 5
> Date: Thu, 11 Oct 2007 13:34:44 +0200
> From: "Rene van Weert" <rvweert at gmail.com>
> Subject: Re: [Asterisk-video] Pach for q931 in misdn
> To: "Development discussion of video media support in Asterisk"
> <asterisk-video at lists.digium.com>
> Message-ID:
> <7e04f3c60710110434o134b0670s950dbbbad7583536 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi, I am using mISDN too. Has anybody succeeded already in making this
> patch?
>
> Cheers,
> Ren?
>
>
> 2007/9/14, Klaus Darilion <klaus.mailinglists at pernau.at>:
>>
>> Sergio Garcia Murillo wrote:
>>> Hi klaus,
>>>
>>> I've recieved a mail from Philippe asking me if it could be possible to
>>> use h324m_call with misdn.
>>> As you had to do a modification to the q931 negotiation for chan_zap, do
>>> you think it could be
>>> possible to make the same patch for misnd?
>>
>>
>> I have no clue about misdn but I guess it should be easy to add. It is
>> only a certain transfer capability and a certain user information layer
>> 1 value.
>>
>> Thus, by looking at my patch it should be easy to port it to misdn.
>>
>> regards
>> klaus
>>
>> _______________________________________________
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> End of asterisk-video Digest, Vol 18, Issue 10
> **********************************************
WALTER RODRIGUES FILHO
( SIP:WALTER at SIPTEL.NET.BR
* WALTER at SIPTEL.NET.BR
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