If one would visit with knowledgeable transmission engineers that work
full time in the telephone industry, one would find telephony standards
that govern exact transmission levels at each point throughout a
country's telephone network (including the long distance facilities, pbx
trunk loss, CO switch loss, etc). The only variable in those standards
are the end user loops, which varies due to the length of the loop and
other mostly uncontrollable and/or variable factors. The individual
telephone companies oftentimes have internal transmission standards that
govern what is or is not acceptable in terms of end user pstn loops.
Practically all US telcos of any size force their installers to measure
the transmission loss for every new installation, and oftentimes on any
repair call.
Asterisk's pc-based analog I/O cards totally ignores those standards.
So, an automatic gain control would be nice but it would really be a
work around for other root-cause / design problems.
In testing various analog pstn I/O cards, I've found the sangoma A200D
card (with hardware echo canceler) to be the best pstn analog interface
on the market that address both the echo and transmission level issues
for the longer higher-loss pstn loops. Transmission levels are still a
little bit low but very usable.
JD Austin wrote:
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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