Hi all,
This is my first question to this list, so please be gentle...
Last week I installed a X100P FXO-card. And with not much tweaking I had
it running fine, there is only one problem right now and it is the
soundquality.
When making a call sound is always perfect, both for the calling party
Hello Jim,
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
BTW: Whats actually that SendDTMF ? thing ?
Thx for the
Hi,
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is, replace it with what?
The idea we have been playing with was to get a Digium E1 card (we
already have bought lot
On Mon, 2005-02-28 at 09:58 +0200, Mark Kidd wrote:
Hi all
i need urgent help our entire switchboard is down only 5 days after it came
up.
this is the second time this has happened and i am thinking that asterisk is
not worth the trouble it gives.
Or you don't know enough about asterisk
On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote:
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
Pray tell me why. The list isn't being flooded by these messages as far
as I see.
--
Dave Cotton [EMAIL
On Mon, Feb 28, 2005 at 12:35:29AM -0330, Paul Fielding wrote:
You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from the previous
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is, replace it with what?
The idea we have been playing with was to get
I would like connect two offices where one office have 4 PSTN Analog lines
and another office without any PSTN. Both the offices will have two separate
Asterisk server with TDM400P cards (4 ports FXS FXO).
My questions is that how to configure Asterisk to forward the PSTN calls
directly
to
Hi all
Ihave registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones.
i configured the extensions.conf file in both the server.
the extensions.conf file on server 192.168.0.9 is
exten=301,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
Hi all
Ihave registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones.
i configured the extensions.conf file in both the server.
the extensions.conf file on server 192.168.0.9 is
exten=301,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
On Mon, 2005-02-28 at 20:38, Azhar Chowdhury wrote:
I would like connect two offices where one office have 4 PSTN Analog lines
and another office without any PSTN. Both the offices will have two separate
Asterisk server with TDM400P cards (4 ports FXS FXO).
My questions is that how to
CAPS LOCK fUnNy
On Sun, 2005-02-27 at 20:25, Roy Sigurd Karlsbakk wrote:
HELP NEEDED TURNING OFF THE cAPS lOCK KEY
:)
On Feb 25, 2005, at 20:07, Edward Banfa wrote:
Hello all,
Hi I would like to know how to configure a Mediatrix 1102 box to work
with my asterisk box. I have analog
On Mon, Feb 28, 2005 at 09:58:28AM +0200, Mark Kidd wrote:
Hi all
i need urgent help our entire switchboard is down only 5 days after it came
up.
Read the other email first, you seem to need to know a little more
about linux also. In any case I do have one hint for you:
[EMAIL PROTECTED]
Hi Howard,
Thanks for quick reply. Although I am searching the mailing and googling, do
you
have a URL about to setup Asterisk with similar situation?
Thanking you,
Azhar Chowdhury
- Original Message -
From: Howard Lowndes [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
On Sun, Feb 27, 2005 at 05:32:49PM -0800, Lee Howard wrote:
Quite right. I'm sorry to have misled.
What happens is this (as an example scenario):
The receiver will, for an example, receive the post-page message. The
sender expects a response to this. The receiver, however, is required
I have * configured with 2 X100P cards (fxs_ks). The lines from the
telco are 'analogue both way ddi trunks'. This means that every inbound
call contains digits that represent an extension on the PBX. I can make
outbound calls from * with no problem however I cannot receive inbound
calls on these
1) Get a 4-port TDM card and install it into your Asterisk box.
Connect the TDM ports to your modem ports. Then forward incoming
calls on fax DIDs to those TDM ports.
Digium TDM 4 fxs is not really a good choice for a faxing system. I've
tested it for a while.
You should read old messages
Hi Mark,
On Mon, 28 Feb 2005, Mark Kidd wrote:
modprobe zaptel - no problems
[EMAIL PROTECTED] root]# modprobe wcfxo
I'm just curious, did 'modprobe wcfxo' ever work? I seem to remember that
for the TDM400P suite, the module to load was (rather confusingly)
'wcfxs', even
On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote:
On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote:
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
Pray tell me why. The list isn't being
Hi,
Its now up at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
I would be interested in any feedback. Hope it helps.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius
Kidubuka
Sent: 28 February 2005 04:50
To: C.
Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header application/dtmf-relay
and you have
On Sun, Feb 27, 2005 at 10:32:21PM -0600, Steven Critchfield wrote:
On Mon, 2005-02-28 at 00:43 +0100, Ilija Poznic wrote:
Hello my name is Ilija Poznic and I have a problem.
My configuration is
1. Digium TDM4000P with one FXS.
2. AVM Fritz ISDN adapter
On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote:
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is,
Martijn van Oosterhout wrote:
On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote:
On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote:
This is getting VERY annoying.
Is there anyone in here that has access to the list administration to
delete the user below???
Pray tell me why. The
BTW: Whats actually that SendDTMF ? thing ?
http://www.voip-info.org/wiki-Asterisk+cmd+sendDTMF
DTMF definition : http://en.wikipedia.org/wiki/DTMF
N.B.: please try to trim your answers, the message is becoming pretty long
hth
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please, all listas.asteriskbrasil.org mailinglist to reconfigure to new IP
addresses sen mail to [EMAIL PROTECTED]
regards,
Max Rivera
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Oups I shouldn't have left that much voice messages those last weeks ;-)
I once got to talk with someone from voiceconduits via AIM, but that's
all, no reply to emails and voicemail!
Marc
ross jones wrote:
Does any one know what happened with voice conduits? I have been trying to
reach them
Russell Bryant schrieb:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greetings Everyone!
Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been
released. There is also a new tarball for Asterisk-sounds.
They are available for download on the digium FTP site:
Asterisk stable still has the old capability of 'sipfriends' and
'iaxfriends' for putting data into MySQL for peers. This is what the
changelog note is referring to. If you need more information on either of
the above, feel free to browse the voip-info.org website! Have a great day.
- Joshua
Azhar Chowdhury [EMAIL PROTECTED] writes:
I would like connect two offices where one office have 4 PSTN Analog
lines and another office without any PSTN. Both the offices will
have two separate Asterisk server with TDM400P cards (4 ports FXS
FXO).
What I would do is use DUNDi. There's an
Hi
Is there a add-on for asterisk where I can define a
rate plan for outgoing international calls and let my sip users make calls
depending on the credit they have.
tks
Kanishka
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Thanks for the great job plus all the others that contributed to this.
I'll certainly use it and give you feedback.
Hi,
Its now up at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
I would be interested in any feedback. Hope it helps.
C
-Original
I got a very strange problem with call-hold
function.
For calls that come in from PSTN and route to a SIP
extension. If I put the call on hold, I cannot unhold the call
after. The caller would be left with hold music forever. A warning
message would be shown on the console usually a few
hi
I'm trying to make video work over SIP between two softphones
I can get audio, but video fails
sip debug is here
http://karlsbakk.net/videotest.log.gz
can someone take a look at it, please?
roy
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I am thinking about a making a web based directory that dials a number with
one click.
From an overview picture does the below look like the correct way to go
about it:
web app sends something like the below call file to asterisk
Action: Originate
Channel: SIP/1010
Context: demo
Exten: 1234
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote:
sendfax (and mgetty) requires a modem interface. The zaptel interfaces are
raw tdm interfaces. SpanDSP could be made to provide a smartmodem
interface but no such code exists
Hi Roy,
Did you check the video codec on the EyeBeam side ? I think that * works
properly only with basic h263.
Btw, to start video you have to push manually the start video button
(ok, that sounds silly but it's not that intuitive...).
We have tested it with no nat, and it works fine in those
Hi,
I wonder if I can securely authenticate two Asterisk servers with IAX
connection. I know for RSA authentication for IAX2 channel, but that seems
to be meant for peer authentication...
Has anyone done RSA (or any other secure way) authentication between two
Asterisk servers ? Any example ?
Hello.
I found out about asterisk a few days ago looking for an alternative voip
solution to cisco and lucent (they have very expensive solutions).
The question is... the company works with 2 E1 incoming lines that go directly
to 50 call center agents and the rest must be redirected to other
1) Get a 4-port TDM card and install it into your Asterisk box.
Connect the TDM ports to your modem ports. Then forward incoming
calls on fax DIDs to those TDM ports.
Digium TDM 4 fxs is not really a good choice for a faxing system. I've
tested it for a while.
You should read old messages here
sorry test.
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Lee Howard wrote:
On 2005.02.27 11:28 Jon Gabrielson wrote:
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial
Lee Howard wrote:
On 2005.02.27 09:30 Martijn van Oosterhout wrote:
On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote:
Fax cannot handle a one-second delay. As Steve mentions in the
article, per-spec fax has some timings (particularly silence in
direction switching) set at 75 ms +/-
Title: Fax Failing
Hello All,
I am trying to set up faxing using [EMAIL PROTECTED] 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI
Peter Svensson wrote:
On Mon, 28 Feb 2005, Edwin Groothuis wrote:
For the project I've used the Eicon DIVA card. It has 8 BRI ports,
and for about 25% of the time there are 7 or 8 in use. So we want
to replace it with an E1 card. Only issue is, replace it with what?
The idea we have been
ross jones wrote:
Does any one know what happened with voice conduits? I have been trying to
reach them for nearly three weeks now. Their voice mail boxes are full and
writing email to them does not get any returns. Thoughts or sightings are
appreciated.
There was a thread a month or two ago
Hello,
I recently downloaded phpconfig from
http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig?login=2
but on installing it, my interface does not look like the one at
http://rd.it.utah.edu/phpconfig/.
The main differences are:
1)On opening a file for editing, on the left menu mine has
w fm3 wrote:
on CISCO 79xx the only way to do it is setup a new line that autoanswers
on the phone and configure each phone to do this manually. - Is this
still correct?
There was a recent post, look at lists.digium.com for the archives, that
detailed an 'auto-answer' script for the 79XX
Questions:
1) Am i using an older version? If so, where can i get a newr version?
2) Am i missing some configuration, which one?
See this newly created document, it explains everything you need to
make it work.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig
It's been
Hi,
i am getting the follwing messages with asterisk 1.0.5
[...]
Feb 28 16:13:05 VERBOSE[8899]: !! Unable to handle ROSE operation 34
[...]
Can anybody gibe me a hint what is is about ?
Greetings,
Martin
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So are you saying that in my setup where I have a
adit 600 channel bank with FXO/FXS connected to a t110p,
that asterisk does an analog bridge? Presumably that
would mean 56k modems, etc.. would also work fine.
I was under the impression that asterisk used iax2 for the
internal trunk.
Jon.
On
help
I just want a list of commands, if this mail shows in the list,
sorry, my bad.
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Hei!
Does anyone know how to configure this phone to autodial the number
after interdigit timeout has passed?
Rennes
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On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote:
Hello Mark , C. All , Is this device available for sale
in the US ? All the digging I've only found outside US
mentions of sales . Any help appreciated . JimL
No idea. The Unit I have is a locally
looks good. did you install the tif to pdf stuff?
(type help-aah for help on how to do this)
--- Wiley Siler [EMAIL PROTECTED] wrote:
Hello All,
I am trying to set up faxing using [EMAIL PROTECTED]
0.6. I have followed
the instructions to the best of my knowledge. When
a fax comes in,
Hello,
That's the document i read and got all the relevant links.
I also tried to follow all the predures .
More help is appreciated,
Thanks very much
Allan
On Mon, 28 Feb 2005 10:13:02 -0500, Time Bandit [EMAIL PROTECTED] wrote:
Questions:
1) Am i using an older version? If so, where can
Try the snom soft phone! http://snom.com
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dave Chase
Sent: Saturday, February 26, 2005 12:31 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
http://www.psitek.co.za/gsm.html
These guys are also in RSA, and Australia.
This unit does exactly the same as the DigiCell,
which mark is talking about, but is a much better
product (and more expensive)
maybe they export ?
-Herman
On Mon, 2005-02-28 at 17:21, Mark Elkins wrote:
On Fri,
Rennes Neps wrote:
Hei!
Does anyone know how to configure this phone to autodial the number
after interdigit timeout has passed?
It's documented on the SIPura web site and the various documentation
for other SIPura products. However, with a proper dialplan in the
phone you seldom need to deal
I thought so. I ran install-pdf from the command line after installing
everything else.
Did I miss something?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, February 28, 2005 8:23 AM
To: Asterisk Users
Mateo,
Dialing the extension to your softphone is the same as any hardware
extension.
Exten = 1000,1,Dial,(SIP/1000,20,trf) pretty
exten = 1000,2,Macro(vmessage,1000)
exten = 1000,3,Hangup
Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the
On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote:
http://www.psitek.co.za/gsm.html
These guys are also in RSA, and Australia.
This unit does exactly the same as the DigiCell,
which mark is talking about, but is a much better
product (and more expensive)
maybe they export ?
On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling
On Mon, 2005-02-28 at 09:15 -0600, Jon Gabrielson wrote:
So are you saying that in my setup where I have a
adit 600 channel bank with FXO/FXS connected to a t110p,
that asterisk does an analog bridge? Presumably that
would mean 56k modems, etc.. would also work fine.
I was under the
Hello,
I am using Asterisk stable and have a question about the queue_log. It
seems like in the past (although I can't find my old logs) that the
exitwithkey produced a wait time entry. It would seem logical that you
would want to track this. Right now it only shows the key they
pressed, and the
Hello,
I've got problems to install zaptel on a SuSE 9.1 System. The System has
got a Linux 2.6.9 Kernel.
If I try to load zaptel framework (modprobe zaptel) I get this message:
FATAL: Error inserting zaptel
(/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
How I can fix this.
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call
I have a desire to incorporate asterisk into some of my network
monitoring. I would like to use the outgoing calling features to
connect a phone (on-call cell phones) to an agi script which can
provide some information to the called party.
Ultimately I would like to pass 2 pieces of information
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
leandro_tenorio
Sent: Sunday, February 27, 2005 8:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] IAX2 (Stupid question)
at least 4 me.
Anyone knows what
Hi,
Does anyone have any experience connecting Asterisk to a Meridian
system using an ATA2 and x101p?
The basics work -- I can make outbound calls, receive inbound, and use
flash to transfer calls, but certain things do not work, specifically
with calls from internal extensions.
- Does the
Hello,
I'm trying to install CAPI Driver for Suse
9.2 and I found the documentation for this pretty old since It refers
toSuse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install).This
is especially apparent when I look at the section of these instructions for
Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the specifics you are using.
Update the settings in your softphone to register the name and number as
1000
Then any attempt to dial 1000 should come to that phone.
Wiley
After doing thoses
How can I make sure that I only connect to a peer once?
E.g., I want that all my staff only use one sipgate connection to dial
out (although I am sure sipgate would love to make more money and let
all my staff call out)?
bye
Ronald
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[EMAIL PROTECTED] wrote:
How do you make the page
http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi
1. use a virtual domain for it, which you do not broadcast
2. use a different name for astcc-admin/
3. limit it to known IP address from where you log in
4. use .httaccess or httpd.conf
Bastian Schern wrote:
Hello,
I've got problems to install zaptel on a SuSE 9.1 System. The System has
got a Linux 2.6.9 Kernel.
If I try to load zaptel framework (modprobe zaptel) I get this message:
FATAL: Error inserting zaptel
(/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
There are a number of models similar to this, they generally go under
the name of 'fixed cellular terminals.'
Most of the gsm cell manufacturers make them...
for example, nokia makes the noia 22 and 32 models (the 22 is hard to get now)
Eurotech have some cheap models using wavecom gsm modules,
I am using the Manager tooriginate calls. I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.
How can I reliably know if the phone on the other end of the line is receiving the call?
Thanks, Tom
Do you Yahoo!?
Yahoo! Mail - 250MB free storage. Do
I fear that list digest did not forward to me all the messages...
buying cell phone adapters is quite unfeasible at this point, since the
installation at hand uses 8 BRI for outgoing calls, and the customer
negotiated very special rates for handling all the traffic through his voice
carrier.
I am getting "Message: Originate failed" even the
phone is ringing on the other end of the line.
Originate will ring your own
extension first and when you pick up, call the other number. If you don't
pick up your extension, you will receive the message you see.
Bill Seddon
From: [EMAIL
Hi,
The default install from that turorial gave me fully functioning links etc.
What format are your config files in; care to post an extract?
What version of PHP are you running?
Regards,
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Allan hank
Hello,
I would like to know how to maximize the number of simultatenous outgoing calls. The application I am working on uses the Manager API to originatethe outgoing calls, and the callswill hang up one or two seconds after the callee picks up the phone. I know it sounds strange but that is what
--On Monday, February 28, 2005 08:46 -0800 beonice [EMAIL PROTECTED]
wrote:
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
Hello,
I'm currently working on a new installation and wondering which
architecture and protocol I should use...
I want to share my Asterisk server between users on my internal LAN and
a user connecting via Internet...
So, my server has to be reachable from outside and also from inside...
For
I wonder if it is possible to setup exensions.conf so, that incoming
calls are charged.
bye
Ronald
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Hi
Does anyone know the procedure for
installing the ring state patch for snom phones . I really need
this.
Id appreciate any
help.
Geoffrey Sachs
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I'd like to find a way to have my asterisk server in a DMZ protected
from outside and not directly on the internal network. Is there any
recommended architecture ?
One of my current installs is a DMZ with an * server protected from outside
and inside with Monowall:
http://www.m0n0.ch/wall/
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote:
Oh I'm sorry. This is the first list I've joined where this
is such a big
issue! Forgive me for not having your superior
understanding of mail
clients,
see bottom
- Original Message -
From: John Millican [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26
Hey Thanks guys...
But how can I use Asterisk for billing and accounting?
Do you mean use the astcc module..?
Please help...
Thanks,
Neel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang
Sent: Saturday, February 26, 2005 11:50 PM
To: [EMAIL
On Mon, Feb 28, 2005 at 05:06:47PM -, Victor Alvarez wrote:
Hello,
I'm trying to install CAPI Driver for Suse 9.2 and I found the
documentation for this pretty old since It refers to Suse 8.2 (
http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This
is
it depends what you mean by billing and accounting. postpaid? prepaid?
integrated into the dialplan or just for use later?
you can use cdr_mysql or similar to dump everything into a DB and
build billing apps on that, if you want as well.
please read the stuff here:
Kristian Kielhofner schrieb:
Bastian Schern wrote:
Hello,
I've got problems to install zaptel on a SuSE 9.1 System. The System
has got a Linux 2.6.9 Kernel.
If I try to load zaptel framework (modprobe zaptel) I get this message:
FATAL: Error inserting zaptel
Thanks for your help. From what you said it looks like
I should not use Originate, but there is no
alternative to the Originate action if I just want
to make an outgoing call is there?
This is what my code is sending to the Manager API:
clientSocket.Send(Encoding.ASCII.GetBytes(Action:
David Brodbeck wrote:
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote:
Oh I'm sorry. This is the first list I've joined where this
is such a big
issue! Forgive me for not having your superior
understanding of
The key to getting the menu entries to appear on the pages is the
fgetc/fgets edits. It caught me out until I read through the code.
BTW - how did that error get into CVS anyway!!
Take another look at the tutorial again to see if you have missed anything
else.
--
No virus found in this
On Monday February 28 2005 1:17 pm, Roger Hanson wrote:
see bottom
snip
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get
the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL
I've been poking at setting up a proof-of-concept * server
as a replacement for our commercial conferencing solution.
I've been through the wiki and list archives, and think
I have found a combination that provides the features we
want/need.
The combination of applications CBMysql and MeetMe2
I'm trying to formulate a strategy for our interconnected Asterisk IAX peers
to failover to the PSTN in the event of a DDoS. We currently use them like
this:
DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP
This works fine, and everyone is happy. One of my concerns, though, is if we
Colin Anderson wrote:
I'm trying to formulate a strategy for our interconnected Asterisk IAX peers
to failover to the PSTN in the event of a DDoS. We currently use them like
this:
DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP
This works fine, and everyone is happy. One of my
Primary * box detects DD0S - runs:
asterisk -rx database put PANIC DDOS YES
and have your dialplan look for that database family/key being set to
determine which path it takes.
When the primary * box detects that the DD0S is over - runs:
asterisk -rx database del PANIC DDOS
On Tue,
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