[Asterisk-Users] PCI front mount chassis?

2004-03-11 Thread Jeff Stohl
I am looking for a good case to house my Digium PCI cards, I was hoping to 
mount them in the front for cleaner access then in the back. Unfortunately I 
haven't found much, does anyone have a good recommendation for chassis to 
use up to six digium cards?

Thanks!

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Re: [Asterisk-Users] PCI front mount chassis?

2004-03-11 Thread Jeff Stohl
I am running six reliably right now.

Surely I am not the only one doing a large capacity single site?


From: Steven Critchfield [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PCI front mount chassis?
Date: Fri, 12 Mar 2004 00:35:40 -0600
On Thu, 2004-03-11 at 23:03, Jeff Stohl wrote:
 I am looking for a good case to house my Digium PCI cards, I was hoping 
to
 mount them in the front for cleaner access then in the back. 
Unfortunately I
 haven't found much, does anyone have a good recommendation for chassis 
to
 use up to six digium cards?

What makes you think you could reliably run 6 cards at once?

The PCI bus only has 4 interrupt lines. Anything over that is shared.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Echo cancellation

2004-02-20 Thread Jeff Stohl
I could kiss you, this resolved my entire echo problem with the X100P.

I have a bottle of scotch with your name on it.

Jeff


From: dkwok [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Echo cancellation
Date: Fri, 20 Feb 2004 12:43:59 +0100
I had problem with echo on my GS phone and it is now resolved. What happens 
was that the order of settings were incorrect in the zapata.conf.

All the echo cancellation settings and other settings to effect on a 
particular channel have to be set out before the occurence of the channel 
no.

see zapata.conf below:
; Zapata telephony interface
;
; Configuration file
[channels]
;
language=en
context=incoming
amaflags=default
cancallforward=yes
callwaiting=no
busydetect=no
callprogress=no
usecallingpres=yes
musiconhold=classic
rxgain=0.0
txgain=0.0
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
usecallerid=yes
callerid=asreceived
callwaitingcallerid=yes
threewaycalling=yes
signalling=fxs_ks
callreturn=yes
channel=1
After the change the voice quality of GS is superior I think it is value 
for money. I would not spend my money on snom or other phones. I would 
consider cisco 7960 is money is not an object though.

--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
 smime.p7s 
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Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.

2004-02-14 Thread Jeff Stohl
I've replaced the cable, no such luck.

Thanks,
Jeff
From: Deepakumar JV [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.
Date: Fri, 13 Feb 2004 07:39:33 -
I had similar problems and finally it turned out to be a telephone cable
problem that connected from the wallmount to X100P card. But with the same
cable i did not have any problem connecting a handset directly (no echo, it
was crystal clear).
May be try replacing the cable with a good one.

Deepak
- Original Message -
From: Jeff Stohl [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, February 13, 2004 01:48 AM
Subject: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.
 I have a basic x100p setup and several soft and hard phones that work
great
 until they hit the PSTN. Like a lot of the posts I've seen I've gone
through
 just about every echo can including mark 2, 3, and the steves with the
 aggressive protection on mark 2. I am running the latest CVS source code
 that I downloaded late last night, 0.7.2.

 My echo is really bad on the dialing out line only, the other user has a
 perfect reception. The inside user has just a horrible echo back of 
their
 voice. I've tried various echo canel methods in zapata.conf as well as
 regular TX/RX gains. I tried running ztmonitor 1 -v and the ztmonitor RX
and
 TX charts came up but did not provide any display back so I was unable 
to
 find some better TX/RX. I tried several settings but those didn't help.

 I am configuring asterisk as a test system and was very happy with the
setup
 until I hit the PSTN. I am hoping the T-1 setup will not have the same
 issues.

 Have I missed any steps in reducing my echo?

 Could my X100P card be prone to echo?

 Could my line really be this bad (it's not on a regular analog phone)?

 Can anyone recommend better echo canceling phones?

 I know this is a frequent topic but I truely have read through all of 
the
 past posts from the December echo can upgrade till today.

 Thanks,
 Jeff

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Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.

2004-02-14 Thread Jeff Stohl
I can hear the echo of my own voice as soon as the X100P clicks over to the 
POTS line to dial the call so I am positive it is not the end party. There 
is no echo between IP calls across country. As suggested by another user, 
I've replaced the cabling and tried two seperate POTS lines wired into 
seperate jacks.

I spoke with several people in the digium IRC room who had the same problem 
and the long and the short of it seemed to be not to use the X100P and 
either suck it up and do POTS into a channel bank or go straight PRI into a 
T-1.

I just purchased a VoiceTronix 4 port FXO card, I'll report my results on 
that front once complete.

Thanks,
Jeff

From: Tilghman Lesher [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.
Date: Fri, 13 Feb 2004 20:59:15 -0600
On Thursday 12 February 2004 19:48, Jeff Stohl wrote:
 My echo is really bad on the dialing out line only, the other user
 has a perfect reception. The inside user has just a horrible echo
 back of their voice. I've tried various echo canel methods in
 zapata.conf as well as regular TX/RX gains. I tried running ztmonitor
 1 -v and the ztmonitor RX and TX charts came up but did not provide
 any display back so I was unable to find some better TX/RX. I tried
 several settings but those didn't help.
If you're getting echo of your own voice, but the remote is getting a
clear signal, then Asterisk echo cancellation is working properly.  It
is the remote provider not echo cancelling properly.
Think of the echo in this way:  what you're hearing is your voice on
the remote receiver being picked up by the remote microphone and
sent back to you.
You might ask your remote party not to use a speakerphone and to
press the receiver tightly against his/her ear, as this will dampen the
sound detected by the microphone and lessen the echo.
-Tilghman

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[Asterisk-Users] X100P / Echo / ZTMONITOR CAN2,3, etc.

2004-02-12 Thread Jeff Stohl
I have a basic x100p setup and several soft and hard phones that work great 
until they hit the PSTN. Like a lot of the posts I've seen I've gone through 
just about every echo can including mark 2, 3, and the steves with the 
aggressive protection on mark 2. I am running the latest CVS source code 
that I downloaded late last night, 0.7.2.

My echo is really bad on the dialing out line only, the other user has a 
perfect reception. The inside user has just a horrible echo back of their 
voice. I've tried various echo canel methods in zapata.conf as well as 
regular TX/RX gains. I tried running ztmonitor 1 -v and the ztmonitor RX and 
TX charts came up but did not provide any display back so I was unable to 
find some better TX/RX. I tried several settings but those didn't help.

I am configuring asterisk as a test system and was very happy with the setup 
until I hit the PSTN. I am hoping the T-1 setup will not have the same 
issues.

Have I missed any steps in reducing my echo?

Could my X100P card be prone to echo?

Could my line really be this bad (it's not on a regular analog phone)?

Can anyone recommend better echo canceling phones?

I know this is a frequent topic but I truely have read through all of the 
past posts from the December echo can upgrade till today.

Thanks,
Jeff
_
Check out the great features of the new MSN 9 Dial-up, with the MSN Dial-up 
Accelerator. http://click.atdmt.com/AVE/go/onm00200361ave/direct/01/

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