Displaying 20 results from an estimated 3000 matches similar to: "Echo problem on SPA-841"
2005 May 10
0
Re: Sipura 841 and headset (Josiah Bryan)
On Tuesday 10 May 2005 9:45 am, David Masure wrote:
> Hi folks !
>
> I bought two sipura 841 phones. I used to have GN Netcom headset
which
> I connect instead of the handset. The problem is that I don't have
any
> sound coming out the headset and I can't speak neither !
>
...
>
> Or....can anyone advise me on headset working with the sipura 841 ?
I just use a
2005 May 10
2
Sipura 841 and headset
Hi folks !
I bought two sipura 841 phones. I used to have GN Netcom headset which
I connect instead of the handset. The problem is that I don't have any
sound coming out the headset and I can't speak neither !
I'am located in France and I was wondering if the cabling in the sipura
and in the headset is the same (I mean the order of the cables) or maybe
is there something else to
2005 Jul 07
4
Sipura SPA-841 Volume Oscillation Problem
Hi all,
The problem is on the volume of the voice sent by the SPA-841. I think the
echo cancel algorithm sets a limit to the microphone when detects sounds or
noise from the earphone. This problem generates an oscillation on the voice
volume sent by the phone and even turns it off completely for very little
lapses of time making the communication very uncomfortable. I manage three
different
2010 Dec 08
2
[headset/mic] Volume too low + echo in * (Gilles)
>
> Different brand/model, but similar as they are both el cheapo,
> entry-level headsets. I tried using them on a laptop, and I get
> marginally better microphone output, even with its volume cranked all
> the way up + automatic gain control enabled.
>
> I guess those on-board soundcards by Realtek aren't as good as a
> quality microphones. I'll get a USB headset
2005 Aug 06
3
SPA 841 form SIPURA
Hello,
How good is :SPA 841 form SIPURA.
Thanks
Varun
2005 Sep 30
3
SPA-841 "Decode Latency"?
We're investigating audio quality issues in our system; maybe someone
can help. We're using Asterisk as a basic PBX, with a single PRI on one
side and SIP phones on the other: Sipura SPA-841's.
We're experiencing several audio effects which seem to commonly
correspond to network failures (packet loss, high jitter, etc manifested
as "robot voice", dropouts, periodic
2009 Sep 09
1
Forecast - How to create variables with summary() results parameters
Hi,
I would like to create variables in R containing parameters of
summary(*Forecast
Results*).
Using the following code:
library(forecast)
data <- AirPassengers
xets <- ets(data, model="ZZZ", damped=NULL)
xfor <- forecast(xets,h=12, level=c(80,95))
summary(xfor)
the output is:
Forecast method: ETS(M,A,M)
Model Information:
ETS(M,A,M)
Call:
ets(y = data, model =
2005 Sep 20
3
sipuras 841 bad sound
Hi Guys!
I have a problems with some sipuras 841 and asterisk 1.0.9.
Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
steve's unicall.
Everything compiled fine and in fact I can make and receive calls but I have
a problem with bad sound when the sipuras call the outside E1's lines. I can
listen to the caller without problems but they heard me with a choppy
2005 Jul 31
0
Sipura 841 vs Grandstream GXP2000
Is there a a consensus on which of these is the better phone.I've
personally been using an 841 and have learned to live with its
shortcomings. I now need to recommend some phones for some sites
we're installing. I'm looking at the BT102 for desktops that don't
want/need a headset but need a phone for the higher end users (without
costing the earth).
TIA,
tony
Zero Effort
2005 Sep 02
1
Snom 360 problem
Good day all
I have asterisk on a box with one network card
I have a 2 companies setup on the system.
To keep all apart I binded a different ip to the interface,i,o,w eth0
192.168.0.254 and eth0:1 192.168.1.254
And in sip.conf I took the bind setting out
So each company's phones are on a different ip range,and all worked well
So we decide to pull the snom190 out and exchange it with a snom360
2009 Aug 01
1
SNOM Phones Displays NR Frequently
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code: snom190-SIP 3.56m
snom320-SIP - snom320 jffs2 v3.36
snom300-SIP - snom300-SIP 6.5.2
Asterisk version - Asterisk
2006 Jun 21
1
SPA-2002 call HANGUP. May be a SIP bug.
Hello,
We have problems with Asterisk and Sipura SPA-2002.
SPA is behind the NAT. Asterisk has nat=yes.
Sometimes call doesn't hangup when user finish the call and hangup the
headset.
In this case during all conversation SIP packets contains
Call-ID: bee522ee-8efa7d25@123.123.123.123
but the final BYE packet from adapter contains
Call-ID: bee522ee-8efa7d25@10.0.0.2
Is such
2006 Jun 28
1
Wiki Voip Phone reviews
Hi,
We have a page on the wiki just for phone reviews, but I think it needs
a bit of format change. Instead of individual reviews for each phone, I
think each person should review all phones they have worked with and
list the phones they have had access to and rank them in relation to
each other. Also each review should have a date so the reader can see
how fresh the data is to current.
2005 Jan 06
0
H.323 to SIP extension
Greetings All-
I have an * box with the NuFone H.323 channel driver installed.
I also have an Altigen VoIP system with a PRI to the PSTN.
I can sucessfully make a call from a SIP extension (snom190)
to an H.323 extension (altigen phone)
The thing I can't seem to make work is a call from a H.323 phone
to a SIP extension.
Here's the layout:
2004 Dec 15
1
Re: Asterisk-Users Digest, Vol 5, Issue 219
I don't think it's the snom, (the break key is set to "off")
the "#" key is not being interpereted by the PBX as an attempt to
initiate a transfer.
Is this an error in my extensions.conf?
Brian
>
>Message: 4
>Date: Wed, 15 Dec 2004 19:39:39 -0500
>From: Info <info@idatasys.com>
>Subject: Re: [Asterisk-Users] Help with transferring a second call
2004 Dec 15
1
Help with transferring a second call from a snom 190
Hello List-
I'm having a problem getting snom 190 phones to transfer a call to
another local extension.
Here is the scenario:
A call (call1) comes in from the PSTN to (exten1). (via pri, if that
matters)
Another call (call2) comes in to (exten1).
(call1) is put on hold while (call2) is answered.
(call2) is then transferred to (exten2) using the "Xfer" button on the
snom phone. This
2005 Jan 02
1
ArtDio IPF-2000 or Sipura SPA-841
I am looking at some lower cost phone to use with Asterisk. What is the
ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has
an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find
any information on it.
Adi
2005 Jan 31
2
SPA-841 Call Waiting
Am I doing something wrong here? GOt a SPC-841 the other day and have it
registering properly. Can place and recive calls as expected but when on
the phone, a second call is immediately dumped to "busy" voicemail. Does
this thing not support call-waiting? Or, have I just got my configs
wrong?
Paul
--
Paul A. Dugas Dugas Enterprises, LLC
paul@dugasenterprises.com
2005 Mar 09
1
Paging and Intercom using Sipura SPA-841
I want to implement a one way announcement and paging facility using
Asterisk and Sipura phones. The wiki says Sipura phones only support
Auto Answer using the Call-Info header which is no lone shipped with
asterisk stable since 1.0.4.
I would like to ask if anyone has implemented a similiar facility
using Sipura SPA-841 or any other SIP phones. If I could take a look
at how
2004 Dec 09
5
Sipura SPA-841
Froogle found me one supplier for the SPA-841, not sure I trust them
though. Does this phone even exist yet? Does anyone have any
experience with it? Does anyone know a vendor other than
Atacomm/voipsupply?