I'm trying to use Asterisk with a TDM400P and 2 analog lines, but I'm
having a hard time getting the kind of audio quality that I'd like.
I'm hoping to be able to use SIP phones to make calls through Asterisk
and have the same quality as a regular analog phone connected to the
PSTN. Are my
Get an FXO card with hardware echo cancellation. I use the Sangoma
A20002D (four FXO ports with echo cancellation). It definitely costs
more, but the hardware echo cancellation makes a huge difference in call
quality! Software echo cancellation doesn't really work...
With this card, would you
there that might benefit you in a situation like this. Go look through
your zaptel source
tree for fxotune and see if it cant possibly correct some of the
problem you're having.
Thanks for this suggestion. I ran the test and activated the settings
with fxotune -sand I'll see how it works. I
No it's not. There will be artifacts in any TDMXXX TigerJet Digium analog
card, IMO. These artifacts are mitigated through the black art and dumb luck
of different chassis, local RF interference, different handsets, different
Asterisk version, etc. But you will most likely never get the exact
Welcome to computer telephony. :-)
I'm actually working in the computer telephony field and have been for
the last 10 years, but I deal mainly with T1s and trunk adapters on
RS/6000s. I'm a software person so I don't do a huge amount telephony
configuration, but I have done my share over
card, IMO. These artifacts are mitigated through the black art and dumb luck
of different chassis, local RF interference
This has me curious.would the RF interference you're thinking
about be interference that affects my SIP phone (earpiece, mouthpiece,
etc), or interference that is
Going to AMP, Setup - General - Extension of fax machine for receiving
faxes = disabled *should* disable fax detection by causing it to use a
different branch of the AMP macro's...
I did set it to disabled, but it still called NVFaxDetect() with a
parameter of zero.
This is most likely your upload speed. I have Comcast supposedly with
384KB upload, but I have a hard time using VoIP unless I use a
low-bandwidth codec like GSM. For g711, it's a crap shoot as to
whether it works or not.
I can always hear the other person clearly since I have a ton of
I have a TDM400P with 2 FXO cards and I'm using [EMAIL PROTECTED] 2.8
I noticed that when I place a call to the analog lines from outside,
Asterisk takes a while to actually ring the extension the call is
being sen to.
I've been doing some tests, calling from my cellphone and here is what I
- After the first ring on my cell, Asterisk logs to the CLI that is
has an incoming call
- After the second ring, it kicks off part of the incoming call context
I fixed this by setting:
usecallerid=no
in zapata.conf
I made this change and it helped in that it reduced the number of
rings.
do... In any case, removing the fax detect seems like it should help.
I commented out the line and it works great. Two concerns though:
- I edited the extensions_additional.conf file, but my fear s that it
will get overwritten if I make further changes via AMP or upgrade. Is
there a way to
I just got a TDM400P with 2 FXO cards. I got it all configured and I
can place and receive calls.
I seem to be getting static on the call, mainly when I speak. E.g, if
I call someone, I can hear them just fine, but they would hear static.
Not a lot...more like a constant background hissing
I'm also using Broadvoice and was having a lot of problems with DTMF.
I 'm in Ft. Lauderdale, FL and I was (inadvertently) using the dca
proxy. When I changed it to use the Miami proxy, my DTMF tones
started to work reliably
I had done some digging and found various posts on the internet where
13 matches
Mail list logo
