Displaying 20 results from an estimated 900 matches similar to: "Grandstream 100 sidetone"
2004 Jan 24
2
Subject: Re: Grandstream 100 sidetone
Chris Albertson wrote:
|What firmware version do you have?
program version 1.0.4.39
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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2004 Jun 22
2
sidetone noticeably loud on analog handsets on T100P
Hi guys,
I've run into a problem that I can't figure out on a bunch of handsets I
have running into a Rhino Equipment 24-port FXS channel bank hooked up
to a T100P and running asterisk-0.9.0 and the associated stable Zaptel
release.
The sidetone (your own voice that you hear in your handset, built in for
comfort) is noticeably louder than it should be, and it doesn't seem to
2006 Jun 13
1
echo sidetone grandstream and tdm400p
Hi all,
thanks to the all of you. This list is very interesting also for a newby like me.
My problem: I just setup my first full working asterisk installation with this
config:
1. n.1 GXP-2000
2. n.4 Budgetone 102
3. n.1 TDM400p (3 FXS, 1 FXO)
Everything seems to work fine, but the sidetone... it's really annoying!
We can hear the sidetone only when we call to the outside (PSTN), it
2007 Dec 07
2
Sidetone with Snom 370
Hi all,
I'm not getting any sidetone on my Snom 370. I searched the web and the snom
wiki, but I don't see any place to enable/adjust it. Callers say I sound
great on the other end, but I don't hear myself so it is a little
off-putting. Any suggestions would be appreciated.
On a related note, some times (maybe 1 out of 10 calls) I get the side tone,
but its delayed by a second or
2007 Nov 10
2
sidetone
Hi -
I've got a new install with a Sangoma A200 and a few GXP2000's. When
users are talking over the Sangoma, they get a lot of sidetone (local
echo). Internal calls are fine. Where do I adjust that? I assume
its in zapata.conf somewhere?
thanks
Todd
2004 Jan 15
1
meetme - ztdummy
On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote:
>> I do not have any zaptel hardware on the Asterisk box, I could not have
>> meetme functioning. I did modify the Makefile in zaptel directory on
>> line 168 by including ztdummy as one of the modules to compile in.
try modprobe ztdummy
This works. Should I include this in /etc/asterisk/modules.conf so that
it will
2004 Jan 15
4
meetme without zaptel hardware
I do not have any zaptel hardware on the Asterisk box, I could not have
meetme functioning. I did modify the Makefile in zaptel directory on
line 168 by including ztdummy as one of the modules to compile in.
The error message from the concole:
-- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
2004 Jan 22
3
Grandstream 101
Just got GS 101 phone and plugged into the network.
Got ip setup however, the following problems arise:
1. when dialing an extension, I cannot further send any key tone to
Asterisk.
2. there is no sound coming from the other end.
I have a sip.conf setup for GS:
[General]
disallow=all
allow=ulaw
allow=alaw
[gs]
canreinvite=no
dtmfmode=info
In the GS101 setting
rtp port = 5004
sip port = 5060
2004 Jan 22
1
sidetone issue
I am using GS 101 and as I am new to Ip phone arena. I am finding it a
bit annoying to hear sidetone, especially when both parties are talking
over each other occassionally. In that case, I cannot hear the other
party's conversation.
Is there any way to suppress it?
Is it only GS or it applies to more expensive phone eg Cisco 7960 as well?
--
David Kwok
Iaxtel/FWD # 17001813482 ext
2004 Jan 20
2
Brandwidth for making internet calls
My ADSL connection speed is 512Kb up and 128Kb down.
When making calls from Asterisk to IAX and back to the Asterisk, the
sound is choppy and 20% of voice messages was lost. What is the
production bandwidth requirement per internet call. I understand there
is no guarantee of QoS but at least a benchmark to follow.
--
David Kwok
Iaxtel/FWD # 17001813482
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2004 Jan 14
4
re hardware requirement - asterisk
I have just checked the Openbsd box on the if interface.
fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500
address: 00:02:55:30:54:28
media: Ethernet autoselect (100baseTX full-duplex)
status: active
inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255
inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1
xl0:
2004 Jan 15
2
hardware requirements - asterisk
In relation to voice degradation when having 2 or more connection to
Asterisk.
The comment on the network setup is quite possible.
I am not too familiar with linux. How do I check whether the asterisk
server's nic is running at full-duplex mode.
Does Asterisk use the sound card on the box to do voice processing?
I am running xlite on 2 pc and making calls through iax, FWD and back to
my
2004 Jan 15
2
re: hardware requirement -asterisk
Referring to my previous post about degradation of voice quality when
having more than 2 connection.
The actual route is:
pc xlite -> local asterisk box -> iaxtel -> local asterisk
I have tried out a different situation:
pc xlite -> local asterisk box -> iaxtel
and the second connection
pc xlite -> local asterisk box -> iaxtel -> local asterisk
The same degradation
2001 Jan 20
2
multi user access to 1 data file
I am running v2.0.7 and have set up a network drive for an accounting ledger system. The software is called MYOB and is quite popular in Australia. This is the first time I have to deal with multi user access to 1 data file.
My setup is:
Global
oplocks = yes
socket options = TCP_NODELAY
socket options = IPTOS_LOWDELAY
[MYOB]
path=/home/office/MYOB
force group = office
directory mask = 0770
2004 Jan 15
2
re: hardware requirement asterisk
This is ifconfig on openbsd box:
fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500
I think this output shows that the fxp0 interface is on simplex mode.
The voice degradation I referred was by using xlite soft phone. I open 2
line similtaneously and dial to FWD and back to my incoming extension.
Xlite is runnning on a w2k box with realtek 100M nic in auto mode. I can
2004 Jan 15
1
Voicetronix Openline 4 + asterisk
Any one has documented how-tos for making voicetronix openline 4 to work
with Asterisk.
I have been contacting Australian Digium resellers and Digium cards are
not approved in Australia. So I suppose Australian users are interested
into putting Voicetronix in use.
Any expereience to share will be most appreciated.
David Kwok
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2005 Jun 07
2
Help! Zap echo on bridged calls
I've been going nuts lately trying to get rid of an annoying echo
problem that makes my asterisk server unusable when clients try to call me.
Here's the breakdown of the issue - Hoping that someone can throw me a
clue:
My setup is as such:
Single AMD Athon machine with X100P clone card and voip through multiple
providers .
* Inbound calls through the X100P that do not bridge to
2004 Mar 11
7
asterisk gui client
I have looked at matt's asterisk gui client at sourceforge. I am not a
programmer by trade. The documentation there seems to be a bit lacking.
Has anyone have the experience in installing the gui client and may
perhaps have a how-to document available for sharing.
--
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel/FWD # 17001813482 ext 1002
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2005 Feb 02
2
different IAX ports for different contexts
I have a problem with my asterisk@home installation (configured with
AMP)
My question is this, can you have different ports for different contexts
within IAX?
[Faktortel]
port = 5036 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
allow=all ; Allow all codecs
register => XXXXX:XXXXX@iax.faktotel.com/EXTEN
2005 Oct 06
14
www.openpbx.org
Hello,
What do you think of this project www.openpbx.org ?
Something like ser and openser !
Kinds Regards
Harry
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