On Tue, Aug 16, 2011 at 05:38:37PM -0700, bilal ghayyad wrote:
The current dahdi version is:
PBX-FF*CLI dahdi show version
DAHDI Version: 2.4.1.2 Echo Canceller:
Well, the output of the dahdi_cfg as shown below, it declares there is
invalid argument. But, really I tried to change the
OK, I can buy echo canceller from Digium and how will be installed in the
digium card? Or it is a hardware?
Currently I am reading a message at the consol that Unable to enable the echo
canceller .. does this means that Digium card that I have is not supporting?
This is the output of the
On Tue, Aug 16, 2011 at 05:34:58AM -0700, bilal ghayyad wrote:
OK, I can buy echo canceller from Digium and how will be installed in
the digium card? Or it is a hardware?
If you're using a TDM2400 you can buy a hardware echocan module or HPEC (which
is a proprietary software echocan) if you
The current dahdi version is:
PBX-FF*CLI dahdi show version
DAHDI Version: 2.4.1.2 Echo Canceller:
Well, the output of the dahdi_cfg as shown below, it declares there is invalid
argument. But, really I tried to change the configuration in the systems.conf
from fxoks=1-16 to fxsks=1-16 but did
Hi All;
To overcome the echo problem, what mainly I have to do in the configuration
other than the following line in the system.conf under dahdi directory?
echocanceller=mg2,1-16
1) How can I know if the digium card supporting echo cancellator?
2) If I am getting a message in the consol that
On Sat, 13 Aug 2011, bilal ghayyad wrote:
To overcome the echo problem...
Digium sells 'High Performance Echo Cancellation'
http://www.digium.com/en/products/software/hpec.php
Also, the 'Oslec Echo Canceller'
http://www.rowetel.com/blog/?page_id=454
is supposed to be
Hello Gareth,
echo also appears when making calls with a SIP phone. These are outgoing
calls.
Another site now also gives feedback on echo, telling they sometimes
also have echo on outgoing calls and if they recall right then sometimes
also on incoming calls (coming from a queue).
This
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
calling with the Zoiper softphone, we
Jonas Kellens wrote:
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the
analogue phone + gateway.
I have the same Grandstream GXW 4008 gateway with 5 analoge phones
attached in another environment and there, there are
Routers wont cause echo. In order for them to do so they would have to
store the outbound voice traffic, delay it and then mix it into the
inbound voice.
Telephones inherently cause echo. For domestic calls the audio path is
normally so short that any echo arrives back so quick the human ear
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the analogue
phone + gateway.
It will present it self on the analogue phone when
Hello,
I stated in my first post that both ends hear an echo when one speaks to
the other...
The only place where echo cancellation is being applied is in the
Asterisk server. I have the following in sip.conf :
;-- JITTER BUFFER CONFIGURATION
Thats the jitter buffer. It has no effect on echo.
So you get echo when calling from the softphone to the analogue phone?
What about when one of those calls somewhere else?
What if they call a regular telephone number?
How do you connect in order to send calls to normal phone numbers?
Jonas
Hello,
I did not say that the analogue phone calls the Zoiper softphone or vica
versa.
Calls are made to from the Zoiper to an external number like a cellphone.
Calls are also made from the analogue phone to external numbers like an
international number in Holland...
Jonas.
On 30 June
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on another Telco-network
: echo
Jonas Kellens wrote:
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
--
_
On 30 Jun 2010, at 13:48, Gareth Blades wrote:
By ITSP do you mean a SIP provider?
ITSP: Internet Telephony Service Provider
S
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Hi!
The network setup is :
analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP
-- other networks
Do it step-by-step: Take the Asterisk server out of the equation, i.e.
call the destination directly with your softphone or the Grandstream ATA
and see if that removes the
Jonas Kellens wrote:
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
Thats where I believe the problem lies. You are
Gareth,
multiple users/SIP-accounts use this asterisk server from many
locations. Like I said: in another location with a similar setup, there
are no echo-complaints on received or made calls.
If you say that it has nothing to do with the Cisco-router, I don't
really know what to go looking
Try the SIP phone. If it is better then you might try looking to see if
there are any echo cancelation settings on the softphone or analogue
adapter you can change. Try turning echo cancelation off aswell since if
there are two running they can interfere with each other and make the
situation
Will turning off the jitter buffer affect the quality of the other calls ??
jbenable = no
I must say I'm not really into these jitter-settings in asterisk. I made
jbenable=yes as it can do no harm...
Jonas.
On 06/30/2010 04:24 PM, Gareth Blades wrote:
Try the SIP phone. If it is better
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, June 30, 2010 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Echo problem in VoIP-calls
Will turning off the jitter
Yes if you have a link where there is a lot of jitter it may affect the
call quality. I would try turning it off to see if it cures the problem
and if it does then you can restore the setting and implement a workaround.
Jonas Kellens wrote:
Will turning off the jitter buffer affect the quality
On 12/20/06, Jason Bachman [EMAIL PROTECTED] wrote:
As I understand it, the echo cancelers in Asterisk only work with the
Analog cards (FXS/FXO).
Not true, echo is caused by any number of things in the voice
network, so Asterisk will echo cancel any Zap device. We use it to
cancel ISDN2e and
We followed these instructions in trying to eliminate echo:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/doc
s-html/x1695.html
Our lines come in from the telco in a PRI, then connect to a Tadiran
switch which hands the lines off to Asterisk over a T1 card.
Steve Davies [EMAIL PROTECTED] writes:
Scott Gifford [EMAIL PROTECTED] writes:
[...]
1.5 to 2 seconds. That is a HUGE delay. echo delay is normally
measured in tens or perhaps hundreds of milliseconds, and you are
unlikely to find a software EC that can deal with a 1.5 to 2 second
delay!
Hello,
We're in the process of setting up an Asterisk server, and are having
echo problems. We have a Digium TE110P, and have tried the MG and
MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and
training times, and with both trunk and 1.2 branch versions of
Zaptel, Libpre, and
Hi,
Did you try to increase echotraining ??
echo training = 800 ..
@++
Hello,
We're in the process of setting up an Asterisk server, and are having
echo problems. We have a Digium TE110P, and have tried the MG and
MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and
training
As I understand it, the echo cancelers in Asterisk only work with the
Analog cards (FXS/FXO). If you are getting echo on a digital line,
there is a problem with either a DAC, the T1 clocking, or you are
getting bit errors. You have a Switch in the middle - perhaps the
switch is doing doing
pixiesfr [EMAIL PROTECTED] writes:
Hi,
Did you try to increase echotraining ??
echo training = 800 ..
Yes, I tried 800, 1200, and 2000; none seemed to make any difference.
Thanks!
---Scott.
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Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's
great but, i get some problems with echo in a adsl line.
My system is a TDM440P with 3 FXO Ports and 1 FXS.
Asterisk 1.2.13
Zaptel 1.2.11
Line 1 - Analog
Line 2 - Analog with ADSL
It's installed with two analog lines one of
More than 128ms?
128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but
definitely more than 16ms.
No, 128ms = 1024 taps
Like what sangoma offers.
Ding, Ding, Ding, Ding!
Okay, to be complete in my answers:
No I do not get more than 128ms delay caused by European routing (I
Message-
From: Steve Davies [mailto:[EMAIL PROTECTED]
Sent: Monday, June 19, 2006 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Problem with T411P
More than 128ms?
128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms
On 6/19/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hi Steve,
Thank you for your answers. First of all span 3 is not a satellite link
and no echo occurs when I connect this line to another pbx with HW EC
feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I
have to do something
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to offer any suggestions.
The only requirement is that 'echocancel=yes' is
Kevin P. Fleming wrote:
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to offer any suggestions.
The only requirement
On 6/19/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Kevin P. Fleming wrote:
- Steve Davies [EMAIL PROTECTED] wrote:
:) Now you've defeated me. I imagine that you need to do something to
enable EC on that card, but it is not a card I know, so I'll leave it
to someone who knows the card to
On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Steve Davies wrote:
We have even experienced problems within Europe where providers route
national calls via international routes to save money. This adds
significant latency and makes any echo so heavily delayed that
asterisk cannot remove it.
-Original Message-
From: Steve Davies [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Problem with T411P
On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Steve Davies wrote:
We have even
Steve Davies wrote:
On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Steve Davies wrote:
We have even experienced problems within Europe where providers route
national calls via international routes to save money. This adds
significant latency and makes any echo so heavily delayed that
Hello,
There are 3 PRIs connected to the card each from
different operators. Especially echo occured on span 3 is really annoying.
Configuration files are as follows. Is there something wrong in conf ?
Zapata.conf --
[channels]
context=default
Title: Re: [Asterisk-Users] Echo Problem with T411P
Try setting echocancelwhenbridged=yes. Also, in your zaptel, you only need to define one span as the clock master, so should be like
zaptel.conf--
span = 1,1,0,ccs,hdb3,crc4
bchan = 1-15,17-31
dchan = 16
span
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration files
are as follows. Is there something wrong in conf ?
Have you verified that the provider on
Steve Davies wrote:
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration
files
are as follows. Is there something wrong in conf ?
Have you verified
Look also at AudioFrames setting on your phone.
I read that it needs to match 20ms packet size of Asterisk packets and it
depends from codec you use.
Mimmus
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Asterisk-Users mailing list
To
Have you tried with apic turned off? And, on another note, our system
had bad sound (you might describe it as choppy) with acpi enabled.
Do you have access to a milliwatt test line?
Moj
sdgesa gaeharth wrote:
thanks for the info.
it is not sharing an irq:
0: 59840409 59803082
I have done this but I still get choppy sound and echo on some callsthanksGiovanni Miano [EMAIL PROTECTED] wrote: Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1
Based only on what I see below (from previous posts), it sounds like you
have two separate issues going on: 1) echo, and, 2) choppy sound. Those
should be analyzed as two problems (not one).
You will find plenty of posts in the archives relative to both. In
general terms, the choppy audio
thanks for the info.it is not sharing an irq: 0: 59840409 59803082 IO-APIC-edge timer 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd:usb1 14: 2141851 2143209 IO-APIC-edge ide0 177: 111558 111273 IO-APIC-level aic7xxx 185:
15 0
Well... the next step (for me anyway) would be to use Ethereal on the
asterisk nic interface to ensure the sip/rtp pkts are reasonable (eg, no
dropouts). If those pkts flow consistently in both directions, then
there must be something impacting the wctdm interface.
Do sip to sip calls sound
ended up setting my TX to -4.5 to cut out the choppiness.
Regards,
Mark.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 14, 2006 12:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] echo problem + choppy sound
I have done this but I
Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1 -2...
2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]:
Can you explain why?Giovanni Miano [EMAIL PROTECTED]
wrote:
I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTN
rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]
:I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the
Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote:rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little
I have installed Asterisk and when I hangup the zap channel Asterisk show this message: Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation
Hello everyone,
How come it is possible that when we make a call (sip to pstn) using a
digium tdm04b we have echo, but if we listen that conversation on an other
sip phone with the asterisk zapBarge application the conversation is really
clear and with no echo ?
With the Digium (paid support)
-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 6:14 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone is
disconnected.
To repeat the question from Matt Riddell:
Does he have Stereo Mix selected as a recording source?
We have found the most common cause of a strong echo to
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 5:05 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related
- Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 5:05 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone
List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 5:05 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone
, 2005 5:05 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone
is
disconnected.
To repeat the question from Matt Riddell:
Does he have
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related.
Yes, this would be the logical conclusion, although it is hard to beleive
given what
Rudolf Ladyzhenskii wrote:
Hi, all
I am running asterisk and my friends are running FireFly IAX phone. All
is fine except one of them. When anyone tries to talk to him, tehre is
a real bad echo. It is nothing to do with sound setup.
Is he using a headset or speakers and microphone?
Does
:12 PM
Subject: Re: [Asterisk-Users] Echo problem -- network related?
Rudolf Ladyzhenskii wrote:
Hi, all
I am running asterisk and my friends are running FireFly IAX phone. All
is fine except one of them. When anyone tries to talk to him, tehre is
a real bad echo. It is nothing to do with sound
Hi ,
I am trying to use a telephone Atcom AT323
Both in SIP mode and in IAX mode, I have a lot of echo on a large number of
number called (NOT ALL, it depends on the network I reach)
I see that using in /etc/asterisk/capi.conf
echosquelch=1
;echocancel=1
echotail=64
Everithing is really good
Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm
CG-410 4 FXOs device. Now I just ordered a few Sipura 3000.
With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the
following :
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes (I tried 800 with TDM04B
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is one of the things also affected. Now I'm using TE110 card in my
system. I hope
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the results you're
only stabbing in the dark playing with these things.
by the way I live in Canada and the
halfway through the call, starts loud and gets quit?
Jon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy
Sent: Wednesday, June 08, 2005 10:37 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Echo problem
Ok I tried Digium
I'm sorry all, lines means config lines of code.
Michael D Schelin wrote:
Hi Martin, There was an great post last week about echo. It stated that
the order of the lines matters. It does. The channels must be listed
last for the echo cancel and most other things to work. Rx and TX gain
is
I use Digium TDM400 cards as well. Asterisk's software echo cancellation
sucks. From what I've heard on the IRC channel, you'll never completely
eliminate echo with it. And unfortunately, hardware echo cancellation starts
out at a full T1. They don't seem to have any solution for someone with
-users@lists.digium.com
Subject: Re: [Asterisk-Users] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the results
you're only stabbing in the dark playing
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
at 17:51 -0700, Kris Boutilier wrote:
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote
I have searched for how to locate echo cancelation on SIP clients, but
cant find anything and echocancel=y doesnt seem to have any effect.
Configuration:
CVS-HEAD from last month
iPAQ h5500 with SJPhone (gsm/ulaw/alaw)
Problem description:
When I place or receive a call I hear a faint delayed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Wednesday, 25 May 2005 2:39 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] echo problem
I have searched for how to locate echo cancelation
On Wed, 2005-05-25 at 14:50 +1000, Terry H. Gilsenan wrote:
Have you tried Xten's softphone for the ipaq? I am using it on a 5550 and a
6325 and the quality is as good as a regular phone. It just worked!
I had echo problems with sjphone on the 5550, and I never even tried it on
the 6325
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Wednesday, 25 May 2005 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] echo problem
On Wed, 2005-05-25
I'm running the latest firmware on the SPA-841 and have a problem
with echo.
The echo occurs on all calls (PRI ISDN on a E110p or SIP) and is not
present when I use the SNOM190 phones so I can def. isolate it down
to the SPA-841s. The codec used is g711u and the phones are on their
own
Brian M. Arlinghaus wrote:
I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On
the 7960s, the echo is quite bad. On the TDM ports, it is there, but not
as bad. I have tried setting echo cancellation to various numbers, but
have had no luck.
This began after a HEAD version of
I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On
the 7960s, the echo is quite bad. On the TDM ports, it is there, but not
as bad. I have tried setting echo cancellation to various numbers, but
have had no luck.
This began after a HEAD version of * was
I've got mostly Cisco 7960s and a few Analog phones
on TDM Ports. On the 7960s, the echo is quite bad.
On the TDM ports, it is there, but not as bad. I have
tried setting echo cancellation to various numbers, but
have had no luck.
This began after a HEAD version of * was installed.
Since
I recently installed * on my firewall and that of a relative some miles
away. I route sipphone
(kphone and x-lite) calls from deep within the backbone (two layers of
firewall) on each end to the other. Works fine between @200Mhz pentium
doorstop linux boxes (even w/2.4 kernel).The problem of
My configuration are:
AlcatelOmni PCX ßà1st
Asterisk Server with ZapCard ßàIAX
trunk over Internetß 2nd
Asterisk Server ßà
SIP phone
I have problem with echo in this configuration. But
when I use sip phone or call trough BRI even if I use IAX trunk I have no
problem
Can someone
So are saying that T2240 will gurantee no echo issues? Did you get any
echo issues with a different PC with the same cards and Pstn lines?
snip
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Thanks for that.
Like many I believe * is unusable in production until these echo issues are quoshed are resolved. Lets hope someone takes up the bounty offer.Rich Adamson [EMAIL PROTECTED] wrote:
So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different
could one at least in the case of the fxo/fxs cards just call out one
port and be looped back into the other, record the outgoing and
incomming call (one recording / port) then compare the phase
difference of the 2 recordings?
-Galt
On Fri, 16 Jul 2004 13:28:46 -0600, Rich Adamson [EMAIL
So are saying that T2240 will gurantee no echo issues?Did you get anyecho issueswith a different PC with the same cards and Pstn lines?
Taff.Steve Underwood [EMAIL PROTECTED] wrote:
Rich Adamson wrote:On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery,
John Galt wrote:
could one at least in the case of the fxo/fxs cards just call out one
port and be looped back into the other, record the outgoing and
incomming call (one recording / port) then compare the phase
difference of the 2 recordings?
-Galt
That is probably the simplest way to
After speaking with several people, and even participating in a forum of
several other people with echo issues, I thought I'd share what we've
done (well actually what our chief RD engineer, Brett Bourn has
done...)
First let me say that normal cheapy PC hardware couldn't be made to
function with
After speaking with several people, and even participating in a forum of
several other people with echo issues, I thought I'd share what we've
done (well actually what our chief RD engineer, Brett Bourn has
done...)
First let me say that normal cheapy PC hardware couldn't be made to
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but...
What's
On Jul 16, 2004, at 11:07 AM, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but...
What's common and not so
On Friday 16 July 2004 12:43, W. Kevin Hunt wrote:
First let me say that normal cheapy PC hardware couldn't be made to
function with out echo. We tried on both the single port Digium T1 card
and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/
hardware scsi raid we had echo
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote:
No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b
or x100p running any Head cvs after June 23rd (totally stock install).
Wouldn't necessarily recommend this box for any commercial production
use, but...
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