Func_odbc is your friend.
Check out func_odbc.conf for odbc access from the dialplan
Check out res_odbc.conf to allow you to use odbc as a realtime source
Julian.
Callum McGillivray wrote:
I was hoping for something more along the lines of the Asterisk CMD
MySQL().
I could always resort to
Ahhh - now this seems like the kind of thing that I was after
Where do I define the available DSN's though?
Julian Lyndon-Smith wrote:
Func_odbc is your friend.
Check out func_odbc.conf for odbc access from the dialplan
Check out res_odbc.conf to allow you to use odbc as a realtime source
That would be in your odbc.ini setup (normally /etc/odbc.ini)
Julian
Callum McGillivray wrote:
Ahhh - now this seems like the kind of thing that I was after
Where do I define the available DSN's though?
Julian Lyndon-Smith wrote:
Func_odbc is your friend.
Check out func_odbc.conf for
Did it identify a card?
rmmod wctdm; modprobe wctdm; dmesg | tail
rmmod wctdm; modprobe wctdm; dmesg | tail
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
Errr. What does that mean?
Cameron
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use ulaw/alaw
with them too..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: 22 April 2007 09:18
To: Asterisk Users Mailing List -
Hi,
Microsoft Dynamics CRM 3.0 integration with Asterisk/Trixbox has been
included in StarJunction and Star Outlook Dialer. This is in addition
to existing support for SugarCRM and Salesforce CRM. It is available
at http://www.starutilities.com/staroldialer.htm
Thank you for your valuable
Tzafrir Cohen wrote:
On Sat, Apr 21, 2007 at 08:59:27AM +0100, Senad Jordanovic wrote:
What about creating a configuration file on server for each soft
phone extension automatically and then importing that file into the
soft phone?
In another words, user receives a link to the setup
Hello all,
I have got an asterisk server in my LAN, getting access to internet trought
a router. I have observed in my asterisk box, when the internet connection
in down, the phones can not register to my asterisk. It is like chan_sip,
does not work without an internet connection.
If when the
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a call
through to the first available transport on a list, such as:
I have two SIP
On Mon, 23 Apr 2007, Adrian Marsh wrote:
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use ulaw/alaw
with them too..
ulaw is g711u ...
g711 (u or a), or ulaw or alaw which are the same things will give you the
best audio quality, but it's not
Based on my experience I would say that using ${DIALSTATUS} variable would
be the most common way to do it...
On 4/23/07, Daniel Pittman [EMAIL PROTECTED] wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular
I have tried to set up app_rxfax to receive faxes over IP. I realise
there are mixed stories about how reliable this is at the best of times,
but at this point all I'm after is some guidance in interpreting the log
below. What does RTP: Received packet with bad UDP checksum suggest?
Here is
I have tried to set up app_rxfax to receive faxes over IP. I realise
there are mixed stories about how reliable this is at the best of times,
but at this point all I'm after is some guidance in interpreting the log
below. What does RTP: Received packet with bad UDP checksum suggest?
Here is
hello,
I have two new cards, one is A400P01 from OpenVox and the other is a BILLION
ISDN.
I have Debian Etch installed.
I want install this packages:
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
Right...so u mean this is difficult:
The problem is that it is not. And encourges bad habits.
--
Dear $USER,
The setup program for your soft phone can be downloaded from here:
http://LINK
During the setup you will be asked for configuration file. Please use
attached file.
you need to use apt-get install asterisk.
If you MUST HAVE 1.217 or your cats die, there are repositories available. For
example, read this: http://www.buildserver.net/
If you still MUST build asterisk yourself, I wish you good luck.
On Monday 23 April 2007 13:29, Josu Lazkano Lete wrote:
On Mon, Apr 23, 2007 at 12:29:53PM +0200, Josu Lazkano Lete wrote:
hello,
I have two new cards, one is A400P01 from OpenVox and the other is a BILLION
ISDN.
I have Debian Etch installed.
I want install this packages:
Robert Lister (robl) writes:
transcode out of .gsm?) I am not sure what parts of the system are
enabled/disabled without the licence.
This mentions voicemail g729 in pass-thru mode. I'm not sure if it works as
I've never tried it, but it may be worth a try...
Well g.711 (a/u) is essentially the same as PCM (raw) which is used to
traditional circuit switched voice environments.
There are such mechanisms as packet loss concealment, where a
predictive algorithm is used to determine wave forms.
But outside of this, whether you use g.711 or g.729 you
Hi,
I am trying to achieve 3-way conferencing taking hint from wiki link
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
Here is the scenario:
1. user ua1 calls user ca1
2. ua1 then presses the feature code *0 to redirect ca1 to
conference room 300
3. ua1 then dials the user 33
On Mon, Apr 23, 2007 at 11:12:15AM +0200, voip crazy wrote:
Hello all,
I have got an asterisk server in my LAN, getting access to internet trought
a router. I have observed in my asterisk box, when the internet connection
in down, the phones can not register to my asterisk. It is like
its in a beta state with only one member... is it a stable project?
thanks
On 4/23/07, Lee Howard [EMAIL PROTECTED] wrote:
nik600 wrote:
i have a PRI connected to a TE205P.
Actually, can i send and receive FAX through Asterisk using stable
solutions?
Or shall i connect an ATA to
Hi all,
Here is my configuration: Phone ←→ Asterisk ←→ Gateway (SIP 2 PSTN)
In the Gateway (patton) I have in codec order G729 then G711
If the Phone use G729, I have a pass-thru in the Asterisk ... It's the main
case.
But If I put G711 in the Phone, I want that the Asterisk try a G711 codec
Hi all.
chan_zap not compiling, yes yes I know this sounds trivial but here me
out...
This morning I decided to upgrade to Asterisk 1.4.2 and Zaptel 1.4.1. I
successfully installed zaptel 1.4.1 and the card is picked up and
correctly configured.
ztcfg shows the following:
[EMAIL
On Mon, Apr 23, 2007 at 06:18:32PM +1000, Daniel Pittman wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a call
through
Monday, April 23, 2007, 12:44:08 PM, Diego wrote:
you need to use apt-get install asterisk.
If you MUST HAVE 1.217 or your cats die, there are repositories available.
For
example, read this: http://www.buildserver.net/
If you still MUST build asterisk yourself, I wish you good luck.
And 10,545 downloads in 33 releases. Crazy people!
nik600 wrote:
its in a beta state with only one member... is it a stable project?
thanks
On 4/23/07, Lee Howard [EMAIL PROTECTED] wrote:
nik600 wrote:
i have a PRI connected to a TE205P.
Actually, can i send and receive FAX through
I have to re-image one phone, I do not want to setup a small network
with DHCP and FTP to get it done. Can I just point the phone at the
server manually to try to bypass putting another dhcp server on my
network.
___
--Bandwidth and Colocation provided
Greetings list,
Hoping someone might have experience with poorly-performing net connections and
which devices work best over them.
One of our clients has a number of employees that work from home, and are given
a SIP phone to take with them and hook up to their broadband. For the most
part,
Hello, everyone.
I'm interested in one thing: as I know asterisk retranslates the media
stream with the next way
1. Gets the frame with the UA1's codec
2. Retranslates it to slan
3. Ratranslates slan to UA2's codec
4. Send the frame
It seems to me, that it follows these steps anyway, the question
hello friends, I am configurin my Billion ISDN and when I start asterisk
(asterisk -vvvc) I have this error message:
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal
sign) in
hello, I have the A400P01 from OpenVox.
Is necesary to install all this packages?
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
http://ftp.digium.com/pub/libpri/releases/libpri-1.2.4.tar.gz
or just with asterisk
Hi,
I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers.
I installed the new driver (0.3.1-rc30) on our pbx but since no voice
was passing I decided to go back to old version (0.3.1-rc23).
Last friday everything seemed to work fine but now every incoming
call drops after
Jordan,
After the phone powers up go into the setup menu, before the autoboot, and
set the following:
Under DHCP Menu set Boot Server to Static
Under the Server Menu setup your boot server information.
If you want to completely forgo setting up an FTP server for the files you
might look at
I don't really know what Billion ISDN is, but some basic Asterisk
troubleshooting seems to be in order. What does your zapata look like? Looks
like you have some errors in there...
Next, you have PRI, right? did you compile libpri after installing zaptel?
Finally, you need to make sure your
sorry, i absolutely don't wont to minimize this project, i've just
noticed that it is in a beta state, and i need a stable solutions, for
a business activity.
Can you or someone else give me some feedback?
I know that Fax over Voip doesn't yet have a stable and complete
support, but i have to
can you help me please???
We're in a better position to help if you can post your Zapata.conf
zaptel.conf files for us to take a look.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from
No. My understanding is that codec translation only takes place when the
codecs are not the same OR if asterisk is recording the conversation. (The
second situation may not require conversion either)
On 4/23/07, Alexandr Olekhnovich [EMAIL PROTECTED] wrote:
Hello, everyone.
I'm interested in
Dears
Can anyone guide me ……
I want to put my asterisk system on an iso image like trixbox ,or how to make
a.
how can I do that ,I am using centos 4.4 final
Regards
_
*
No employee or agent is authorized to conclude
Hi, Tzafrir:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
attached file.
I tried this link, but it's
The Digium website has a section that lists systems and motherboards that
are known to have incompatibilities with Digium hardware ( see
http://www.digium.com/en/docs/misc/compatibility_notes.php ).
Is there a _recommended_ hardware list, one that lists server types,
motherboards, chipsets, etc.
Some codecs are more tolerant of packet loss then others, but I don't think
that the type of codec will have a major effect on its ability to deal with
jitter. Jitter buffers will help but with the side effect of increasing the
overall latency of the conversation (hence the buffer). Lost
It's your understanding and mine, but I need to know exactly. It's not easy
to check.
On 4/23/07, Nicholas Campion [EMAIL PROTECTED] wrote:
No. My understanding is that codec translation only takes place when the
codecs are not the same OR if asterisk is recording the conversation. (The
Hi all,
problem solved!
It was a telco problem.
Giorgio Incantalupo
Giorgio Incantalupo wrote:
Hi,
I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers.
I installed the new driver (0.3.1-rc30) on our pbx but since no voice
was passing I decided to go back to old version
On Mon, 23 Apr 2007, Chris Bagnall wrote:
Greetings list,
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home, and
are given a SIP phone to take with them and
Setup a queue with linear and a timeout to drop to voicemail.
Thanks,
Steve Totaro
www.asteriskhelpdesk.com
Daniel Pittman wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a
If I am not mistaking, g711u is ulaw. Ulaw and Alaw are the best since
they are lossless, meaning no compression, they also take up the most
bandwidth. Ulaw is the native codec to traditional T1s in the US and
some other places, ulaw is seen in other parts of the world. They both
consume
There you go, it's no longer beta. That should solve all of your problems.
Should you need testimonials there are plenty of them in the
asterisk-users archives, on various wikis around, and you can surely get
some from the iaxmodem-users list as well.
Lee.
nik600 wrote:
sorry, i
It looks like the section you want to look at is channel.c:set_format (line
2808). My understanding is that chan-nativeformats is set to the format
that the channel was created in (GSM for instance) and fmt is set to the
codec we are trying to accept audio from or write audio to. The important
Hi,
I have spend allot of time searching a solution: We have different SIP
accounts that our Asterisk registers to, for example:
[general]
port=5060
disable=all
allow=[...]
srvlookup=yes
pedantic=no
context=start
language=de
register = 0123456789:[EMAIL PROTECTED]/someExtension
Problem 1:
Steve Totaro wrote:
If I am not mistaking, g711u is ulaw. Ulaw and Alaw are the best
since they are lossless, meaning no
Lossless? Our friends at http://en.wikipedia.org/wiki/Ulaw wouldn't lie. :-)
Steve
___
--Bandwidth and Colocation provided by
Alex,
This would fix only one of 2 problems. By setting the CDR source
variables, I'm still left with the monitor variable and any others that
may come up in the future. There was another email saying I should just
set the monitor file name as well. Since I'm using auto_mon, I'm not
sure that
Thank you very much
On 4/23/07, Nicholas Campion [EMAIL PROTECTED] wrote:
It looks like the section you want to look at is channel.c:set_format(line 2808).
My understanding is that chan-nativeformats is set to the
format that the channel was created in (GSM for instance) and fmt is set to
Hi
Are echo cancellation parameters useful when using the ztdummy driver and
no physical card ?
Thanks
Patrick
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Stephen Bosch wrote:
Hi, Tzafrir:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
attached file.
I
On 4/23/07, Patrick Fortin [EMAIL PROTECTED] wrote:
Are echo cancellation parameters useful when using the ztdummy driver and
no physical card ?
No. The echocan software and hardware only cancel hybrid echo. They
do not cancel acoustic echo that would be generated by voip phones
with bad
On Mon, 2007-04-23 at 18:18 +1000, Daniel Pittman wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a call
through to the
On Mon, 2007-04-23 at 15:25 +0200, Josu Lazkano Lete wrote:
hello, I have the A400P01 from OpenVox.
Is necesary to install all this packages?
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.16.tar.gz
2007/4/19, Noah Miller [EMAIL PROTECTED]:
Hi Diego -
I want to make a SIP trunk between a Cisco 2811 router and a Asterisk.
Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk
2XX). Now I want to configure a trunk so that 2811 users can call *
users. I've been reading a lot
Noah Miller wrote:
Hi Shawn -
We have several Polycom 500/501/601's on both a LAN and at employee
homes.
The problem we are having is if our internet connection goes down the
Local
LAN phones loose their connection to the Asterisk Server.
I've checked everything I can think of but can't
I've compiled ztdummy, following the directions on
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy, and when I try to
modprobe zt :
#modprobe ztdummy
FATAL: Error inserting ztdummy
(/lib/modules/2.6.18-gentoo-r6/misc/ztdummy.ko): Input/output error
FATAL: Error running install command
On 2007-04-23 at 13:40:33, Don Fletcher [EMAIL PROTECTED] wrote:
#modprobe ztdummy
FATAL: Error inserting ztdummy
(/lib/modules/2.6.18-gentoo-r6/misc/ztdummy.ko): Input/output error
FATAL: Error running install command for ztdummy
I haven't been able to find any info on what the I/O
On Mon, Apr 23, 2007 at 02:26:41PM +0200, Jan du Toit wrote:
Hi all.
chan_zap not compiling, yes yes I know this sounds trivial but here me
out...
This morning I decided to upgrade to Asterisk 1.4.2 and Zaptel 1.4.1. I
successfully installed zaptel 1.4.1 and the card is picked up and
Stephen Bosch wrote:
Hi, Tzafrir:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
attached file.
I
The problem has pretty much been there from the beginning. I may
re-arrange cards and see if it happens on one particular channel or if
the problem moves with cards.
Thanks for the response.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
On Mon, 2007-04-23 at 21:31 +0300, Tzafrir Cohen wrote:
A stupid question: how do you see that there is no chan_zap.so ?
Another stupid question: did you rerun make distclean and configure
in the asterisk source directory after installing the zaptel driver?
--Greg
Hermann Wecke wrote:
Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized
showroom in Hyderabad (Andhrapradesh, India)?
Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
How was that a troll? Lazy, perhaps --
On Mon, Apr 23, 2007 at 12:58:39PM -0600, Greg Woods wrote:
On Mon, 2007-04-23 at 21:31 +0300, Tzafrir Cohen wrote:
A stupid question: how do you see that there is no chan_zap.so ?
Another stupid question: did you rerun make distclean and configure
in the asterisk source directory after
KomplettPBX has put together a very nice userguide for Asterisk phones
using the Aastra 9133i phone. Is anyone aware of similar works for
other handsets such as the GXP-2000 or other Aastra phones?
To see what I'm talking about, go to the FAQs section of
http://komplettpbx.com Komplett is to be
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
I am not. The soft phone is not the only software on that computer that
needs cetral configuration.
How do you configure the networking on those computers? The mail
clients? How do you deploy updates?
The fundamental problem, as I
Callum McGillivray wrote:
I was hoping for something more along the lines of the Asterisk CMD
MySQL().
I could always resort to something like that.. but I don't want to run
it on a windows server and I really don't want to go to the bother of
writing FastAGI scripts to make it all happen.
On 4/21/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
What about creating a configuration file on server for each soft phone
extension automatically and then importing that file into the soft phone?
In another words, user receives a link to the setup program and the
configuration file in an
dmesg just says
ztdummy: Unable to register zaptel rtc driver
Thanks
Don
David Olsen wrote:
On 2007-04-23 at 13:40:33, Don Fletcher [EMAIL PROTECTED] wrote:
#modprobe ztdummy
FATAL: Error inserting ztdummy
(/lib/modules/2.6.18-gentoo-r6/misc/ztdummy.ko): Input/output error
FATAL:
Shortly, I'll be purchasing a Sangoma A102. I'm wondering if I should
spring for the hardware echo cancellation circuit or not. Upon
initial implementation, the 2 T1 Ports will be used as a passthrough
as we slowly transition off of a legacy PBX. Eventually, we'll only
be using one of the
I installed Asterisk 1.4.2 on a CentOS 4.4 machine. Everything works
but I noticed that I am missing most of the dialplan CLI commands:
pbxskandiamty2*CLI help dialplan
dialplan show Show dialplan
pbxskandiamty2*CLI
On another machine I have in my office (running
Server A has 4 PRIs coming into it, and provides a SIP connection to
Server B. When we call into server B via a DID from one of Server A's
PRIs, we get crackly sound on recorded playback only (prompts, IVR,
voicemail instructions, etc.), but not on actual live voices once you're
talking.
On 23 Apr 2007, at 21:04, James FitzGibbon wrote:
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
I am not. The soft phone is not the only software on that computer
that
needs cetral configuration.
How do you configure the networking on those computers? The mail
clients? How do you
We have installed two of the Sangoma 2 port cards. Both had echo
cancellation. The cost add on was about $450, not $800. I also had
a single port T1 card without the echo cancellation.
The extra money is worth it to me. Less CPU load and it just plain
works. And if a customer is willing
On 4/23/07, Tom [EMAIL PROTECTED] wrote:
We have installed two of the Sangoma 2 port cards. Both had echo
cancellation. The cost add on was about $450, not $800. I also had
a single port T1 card without the echo cancellation.
The extra money is worth it to me. Less CPU load and it just
Carlos Chavez wrote:
I installed Asterisk 1.4.2 on a CentOS 4.4 machine. Everything works
but I noticed that I am missing most of the dialplan CLI commands:
pbxskandiamty2*CLI help dialplan
dialplan show Show dialplan
pbxskandiamty2*CLI
On another machine I have
Looks interesting.
I could not find out what the difference is between the Free and Pro version
of the Dialer. Can you explain or provide a link?
-Original Message-
From: S. A. Kamran [mailto:[EMAIL PROTECTED]
Sent: Monday, April 23, 2007 2:05 AM
To: asterisk-users@lists.digium.com
Might want to confirm what server address you have declared in your sip.cfg
file (assuming you are using network provisioning for the phones).
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: Eric
echo cancel all the way, any company not including that in first place is
just selling a car without the wheels..
i would see the a card without echo cancel as driving in winter with summer
tires..
On 4/23/07, Erik Anderson [EMAIL PROTECTED] wrote:
On 4/23/07, Tom [EMAIL PROTECTED] wrote:
Can anyone recommend which versions of spandsp, libsupertone,
libunicall and libmfcr2 to use to install Unicall on a Trixbox 2.0
machine?
--
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description:
On Mon, 2007-04-23 at 17:17 -0500, Eric ManxPower Wieling wrote:
perhaps you have writeprotect=yes in extensions.conf
No, both machines have exactly the same dialplan. I am using Realtime
Static to load the dialplan. Basically the only difference between
machines is the Linux
Hi:
Chris Bagnall wrote:
One of our clients has a number of employees that work from home, and
are given a SIP phone to take with them and hook up to their
broadband. For the most part, this works fine, but there are an
increasing number where sound quality is poor (chops in and out,
Diego Iastrubni wrote:
you need to use apt-get install asterisk.
If you MUST HAVE 1.217 or your cats die, there are repositories available.
For
example, read this: http://www.buildserver.net/
If you still MUST build asterisk yourself, I wish you good luck.
This kind of commentary isn't
Carlos Chavez wrote:
On Mon, 2007-04-23 at 18:18 +1000, Daniel Pittman wrote:
G'day.
I am having reasonable success getting Asterisk 1.4.2 running and doing
what I want, but I can't figure out one particular idiom that I want:
There are a few situations where I want to have Asterisk push a
Hi ,
I still can't figure this out.
We have two different network connections with a asterisk servers
each. I want to make calls between asterisk servers. (i.e.) i want to
call from a number in Asterisk server A to a number in Asterisk server
B. I want to implement this using SIP. Will be very
On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration file. Please use
I have installed many Sangoma T1 cards without hardware echo
cancellation. Never had a problem. I have with Digium though.
Thanks,
Steve
Mike Lynchfield wrote:
echo cancel all the way, any company not including that in first place
is just selling a car without the wheels..
i would see the
On Mon, 23 Apr 2007 14:05:55 +0100, Chris Bagnall wrote:
Greetings list,
Hoping someone might have experience with poorly-performing net connections
and which devices work best over them.
One of our clients has a number of employees that work from home, and are
given a SIP phone to take with
On 4/23/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Hi ,
I still can't figure this out.
We have two different network connections with a asterisk servers
each. I want to make calls between asterisk servers. (i.e.) i want to
call from a number in Asterisk server A to a number in Asterisk server
Thanks in advance, I answered the questions.
greetz, Tim
2007/4/24, Erik Anderson [EMAIL PROTECTED]:
On 4/23/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Hi ,
I still can't figure this out.
We have two different network connections with a asterisk servers
each. I want to make calls
Hello list. My name is Eric Kosten, and I am new to Linux and asterisk As
a new user of asterisk and Linux I an having problems to some that might
seem small, but these problems are such that I am not sure ware to look!
I managed to take care of some ownership issues, e.g. sip.conf and
hi folks,
I use Dial(Local/[EMAIL PROTECTED]) to make calls received on my DID number
to ring a local extension. I notice that on 8 out of 10 calls, the audio is
NOT working in the incoming direction (DID provider to asterisk). Local
extension 2055 maps to SIP destination homephone, and if i
khaled,
you might check lamppix or knoppix... they have a remastering scheme
http://lamppix.tinowagner.com/
http://www.knoppix.org/
http://www.wifi.com.ar/english/cdrouter.html
haven't played with it in a while but I did create an iso... that
worked! :)
daveC
Khaled Chehab wrote:
You might as well spend an extra few bucks rather then having headaches from
angry customers in the future.
Astawerks
VoIP Hardware sales and consulting
http://www.astawerks.com
614-495-1400
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Thanks for all the replies. Answering the points raised in turn:
How did you perform the speed tests?
Generally using thinkbroadband.com's speed test java applet.
On the matter of the BitTorrent factor: did you have the users connect
the phone, and only the phone, to the Internet connection?
On 4/23/07, Astawerks [EMAIL PROTECTED] wrote:
You might as well spend an extra few bucks rather then having headaches from
angry customers in the future.
Yah - well in this case, the customers are my co-workers :-) Even
more reason to keep 'em happy.
I've ordered the EC card.
Thanks for
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