[Asterisk-video] Patch 0010217

[Klaus Darilion]

Sergio Garcia Murillo schrieb:
> AST_FRAME_DIGITAL anyone?? jejeje 
> By the way Klaus, is any change on 1.6 that we could use on that issue?

No. I still would favor AST_FORMAT_DIGITAL for "digital" voice calls but 
that would require to patch every ISDN channel driver.

I wonder if outgoing 3G calls work for anyone without setting ALAW 
manually in app_h324m.c.

Last time I tried to find the cause of the problem and it is:
- h324m_call requests a local channel in format ALAW|ULAW
- the local channel traverses Asterisk's preferred codecs list and 
decides to use ULAW (Asterisk preferres ULAW)
- when the zap channel is requested, chan zap detects it is a "digital" 
call and thus does not set a certain codec, but uses the "deflaw" 
setting of the zaptel kernel module - e.g. using PRI+E1 or bristuff the 
deflaw is ALAW.
- now Asterisk has a zap channel with ALAW and a Local channel with ULAW 
and starts transcoding :-(

I think most app_h324m users reside in Europe, thus have a deflaw of 
ALAW. Wouldn't it be better to make ALAW the default in h324m_call, or 
make a command line option to set the "tunnel-codec"?

regards
klaus

> 
> Best regards
> Sergio 
> 
> ----- Original Message -----
> From: Valerio Puglia [mailto:valerio at oscorp.sm]
> To: asterisk-video at lists.digium.com
> Sent: Fri, 14 Mar 2008 10:36:59 +0100
> Subject: Re: [Asterisk-video] Patch 0010217
> 
> hi Klaus
> 
> i remove AST_FORMAT_ULAW and it works
> 
> but when bridge 2 mobile thelephone or call from sipphone to meobile 
> phone the video doesn't start.....
> 
> 
> 
> 
> 
> Klaus Darilion ha scritto:
>> Hi Valerio!
>>
>> I guess it as a codec problem inside asterisk. app_h324m tunnels the 
>> digital call inside G711. Then, sometimes asterisk tries to transcode 
>> from alaw to ulaw.
>>
>> Please search in app_h324m.c for AST_FORMAT_ULAW and remove it (there is 
>> some comments which tell you how to do it), so that h324m_call forces 
>> the usage of ALAW (which is the default of zaptel when using E1).
>>
>> Let me know if this worked for you.
>>
>> regards
>> klaus
>>
>> Valerio Puglia wrote:
>>   
>>> Hi Klaus i have inserted your patch in asterisk 1.4.17 and libpri.. the 
>>> call out work prefect but when arrive the videocall ..and i accept the 
>>> call the telephone remaing in wait (also is resond) but the sip phone 
>>> the call is already upcoming ... asterisk doesn't listen the answer...
>>> i try to SIPPHONE > TO CELL
>>> and bridge 2 mobile phone... but the same result...the celluallar phone 
>>> caller remain to calling state...but the other is waiing for video(like 
>>> as it had answered)
>>>
>>> do you have any idea for my problem?
>>>
>>>     
>>
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> 
> 
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