Displaying 20 results from an estimated 1300 matches similar to: "meetme - ztdummy"
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 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 Jan 24
3
Grandstream 100 sidetone
For people who are using GS 101, what do you think the sidetone
generated by the phone.
I find mind a bit annoying. It has a delay and you notice it as an echo.
The volume of the sidetone is also quite hight. I am distracted when
both caller and called party talking over each other occasssionally.
The volume of the sidetone can be turned down using the volume button
but it also control the
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 16
0
Re: Asterisk-Users digest, Vol 1 #2525 - 11 msgs
Subject: [Asterisk-Users] meetme not working
Reply-To: asterisk-users@lists.digium.com
I am trying to set up meetme functionality but am unsuccessful so far.
When I dial the extension, an announcement says, "That is not a valid
conference number, please try again."
In order for meetme to work you need either zaptel hardware OR you
recompile the asterisk directory and include the
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 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
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 16
0
re: hardware requirement -asterisk
Philipp von Klitzing wrote:
> You'll need to provide the CODEC that you are using in X-Lite!
>
The codec used in Xlite is 711uLaw. I guess it is one of the preferred
ones other than gsm. And it is of small size.
--
David Kwok
FWD#/IAXTEL# : 17001813482 ext 1002
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2004 Jun 11
3
DID/T1
I need clarification as to DID in T1 connection.
T1 provides 24 channels for voice/data. Do it assign each channel to
particular DID. Or you can have unlimited DID to share the 24 channel as
an example. ie. Outgoing/incoming traffic is not bound to particular
channel. Whatever is available will be used according to the grouping in
zapata.conf.
--
David Kwok, CISSP
Tel: 612 82315701 ext 1002
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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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
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 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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2004 Jan 20
1
PSTN Gateway
Hello,
I am looking for information on setting up digium FXO card for use as a
PSTN Gateway (H323-PSTN) to work with GNUGk.
I am basically looking for the setup and it would be great if anyone can
share his experiences with the same. Also, if there are any limitations in
going for such a setup and problems that may arise/things that I should keep
in consideration.
Thanks & Regards,
Deepak
2004 Apr 05
2
iax2 trunk - unable to accept trunk packet
I have 1 * box having x100p installed and the other has no zaptel card
at all. both of the * box has compiled in ztdummy module and both have
been activated by modprobe ztdummy.
When using trunk to connect the 2 *box. The one without zaptel card
complaint about unable to accept trunked packet: no matching peer.
On the one has zaptel card I have tried to remove the ztdummy module and
connect
2004 Feb 17
2
x100p dropping incoming calls
I have been experiencing hung up when answering incoming calls through
x100p.
NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered)..
-- Executing Wait("Zap/1-1","1") in new stack
-- Executing Answer("Zap/1-1","") in new stack
-- Executing DigitTimeout("Zap/1-1"."5") in new stack
-- Set digit timeout to 5
--
2004 Jan 21
1
Zap show channel
What are the meaning of these Zap show channel output?
Caller ID string:
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax handled: no
Pulse phone: no
Echo Cancellation: 0 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0,
2004 Jul 12
2
Oz ISDN
In Australia, Telstra, the local telco provides isdn modem for isdn
connection. The modem has 2 analogue telephone jacks and a serial port
for connection to dialup internet.
My question is that will it be possible to use Zaptel TDM02B to connect
to the analogue jack instead of getting a fritz card to do the
telephony. Will there be less feature if doing so?
--
David Kwok, CISSP
Tel: 612