[asterisk-users] zap channel media volume

[JD Austin]

I've been struggling with this issue for over a year. 
I wish there were some kind of automatic gain control built in to set 
the rx/tx gain on the fly based on the volume of the two channels.
Probably not realistic though.
Is there other hardware other than digium's that better deals with this 
issue?

Rich Adamson wrote:

> The root cause of the low volume problem is the result of software 
> echo cancellation software, and its need to insert a noticeable loss. 
> If I recall correctly, the wctdm.c driver has a statically defined 
> loss value of something like -6 db that is loaded into the TDM400 
> chipset at driver load time.
>
> Ordinarily, that loss is not all that noticeable. But, if your pstn 
> line is rather lengthy (greater then about 5db worth of loss), the two 
> loss values become very noticeable and marginal to users. There is no 
> known fix or workaround.
>
> The low audio becomes even worse when a pstn caller leaves a voicemail 
> and the user calls in via the pstn to retrieve his voicemail. The 
> voicemail gain setting was intended to be sort of a workaround, but 
> its marginal at best.
>
> JD Austin wrote:
>
>> I've been fighting with this issue for over a year.
>> There are several threads here talking about it:
>>    Digium Zaptel volume issues
>>    setting of volume
>>    Low volume/audio problems on TDM400 card
>>    increase the volume ?
>>
>> There is one thread (Voicemail volume adjustment) that give me hope 
>> that this can be fixed that mentions adding
>> |usg(10) to the dial command to increase the gain. I'm still a novice 
>> at the inner workings of asterisk so I'm hoping one of the gurus on 
>> the list will figure this out eventually.
>>
>> JD
>>
>>> Hi all,
>>>
>>> we do have the following configuration
>>>
>>> (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM 
>>> Gateway) -> GSM Enduser
>>>
>>> The call is originated on the (non-Asterisk PBX) - gets send over a 
>>> T1 connection to the asterisk server (which does least cost routing) 
>>> - the asterisk server then does send the call over a GSM Gateway 
>>> into the world...
>>>
>>> The Problem we do have is - that the Users behind the non-Asterisk 
>>> PBX are complaining about low volume media if the the calling 
>>> through the gateway (if the are calling mobiles...). So i have 
>>> started to raise the rxgain value for the connection between the 
>>> asterisk box and the GSM Gateway, this does work quite well - but 
>>> not really perfect. The ringback (not locally generated - does come 
>>> from the GSM Provider) does get terrible loud - as soon as the 
>>> callee is connected - the speech is nearly not hearable because it 
>>> has such a low volume.
>>>
>>> The ringback is EARLY MEDIA - if i am right - and the speech is 
>>> normal MEDIA. So, is it possible to set different gains for EARLY 
>>> MEDIA and normal MEDIA ?
>>>
>>> Does anyone else have had this problem ?
>>
>
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