I have followed all the install note for A2billing and have everything
installed and configured and my asterisk works except the callingcard
application.
Added the following
[callingcard]
; CallingCard application
exten = 777,1,Answer
exten = 777,2,Wait,2
exten = 777,3,DeadAGI,a2billing.php
I guess the obvious question would be whether the callingcard context
is included into the context that the call is coming from. That's the
usual reason for a failure like this.
[EMAIL PROTECTED] wrote:
I have followed all the install note for A2billing and have everything
installed and
Thanks Rob, that helped a little bit but now getting a different kind of error:
-- Executing Dial(SIP/9614-3896, SIP/777|200|rt) in new stack
Feb 18 06:00:30 NOTICE[12236]: app_dial.c:1011 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)
==
On 15 Feb 2007, at 09:55, Yuan LIU wrote:
From: Il Neofita [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however,
is not the case and a really did not catch why.
Now I see where the confusion comes from. Asterisk doesn't
Its not right. I am using a2billing calling card and it works fine
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
If you need to contract Customer Service
Please use our IAX2 WebPhone at the link below
http://www.bochterservices.com/voip/iaxphone.php
nope
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
If you need to contract Customer Service
Please use our IAX2 WebPhone at the link below
http://www.bochterservices.com/voip/iaxphone.php
Rob Hillis wrote:
I guess the obvious question would be whether
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Asterisk recognises them as native uLaw files
From what I know, .ulaw
On Fri, Feb 16, 2007 at 08:19:27AM +1100, Eric Bishop wrote:
Hi all,
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Hi list,
We are looking for an Asterisk consultant for a 3 months mission in Paris. If
you are interested, please contact us at [EMAIL PROTECTED]
Best regards,
Brian
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog interface like FXO
the line has no way of knowing you just hung up and will continue to
ring, which asterisk will see as a new call. in my experience even
when using a PRI if i
Hello,
I'm not familiar with A2billing but for me it is strange that you dial
SIP/777 - 777 should be an extension.
Could you post your user context - or at least the one which direct
you to:
Dial(SIP/9614-3896, SIP/777|200|rt)
Best regards,
## nini @ www.modulo.ro ##
[EMAIL PROTECTED]
a2billing is your default context in the a2billing.conf file from setup,
if you have changed this, and also changed your context for your
card/sip entry to callingcard you should, when you dial from your phone,
be using the callingcard context. Please check your SIP entry for your
phone in
Hi,
I am a retired telephone tech/manager who recently had a bad experience with a
local company offering digital phone service (VoIP). I have spent the last
thirty years in the PSTN network, switching, PBX and key system field and am
interested in learning more about VoIP. My background also
Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet for an answer to this one. My phone company
provides no disconnect supervision. Hence I'm forced to use the
busydetect
feature. I have a TDM400 with two FXO ports. If
I also include a consideration from mine: I would happily use
Trixbox, because I did FreePBX setup once and it was a real pain, but
I'm very frightened by a few issues:
1) Trixbox Macho installation that installs everything without
asking. I, for example, would like to use software RAID (maybe
Go to your book store and get the Fedora/Linux reference.
Get yourself a PC with 20GB drives, a CD burner, and decent ram. The PC
should have either an i386 or x86_64 processor. If you'll be purchasing
a PC, go to the computer store, purchase the piece parts, and assemble
it yourself (I
Original Message-
Ryan McDaniel wrote:
I have a very strange problem I'm hoping someone has encountered
already.
I've scoured the internet for an answer to this one. My phone
company
provides no disconnect supervision. Hence I'm forced to use the
busydetect
feature. I have a
Michael Welter wrote:
Go to your book store and get the Fedora/Linux reference.
Not to start a religious argument here, but it seems from reading the
list that the CentOS flavors have fewer problems
Install Linux. Take all the defaults. Load all packages.
When it's running, login as
Another perspective:
Fedora is not the only linux distro. If I did use it, I would be
building RPM rather than doing local builds on the target asterisk
server. I use debian so I build deb files instead of rpm files.
Here is a good way to get started using debian linux and some cheap old
Hi, Trevor:
Trevor G. Hammonds wrote:
Stephen Bosch wrote:
Are BRI circuits what phone companies call digital lines for use
with digital sets, such as with digital Centrex?
I'm not aware that Telus even offers BRI.
Sorry -- BRI is ISDN, not digital Centrex.
I'm still not aware that Telus
It will do so automatically if it is working. Asterisk will stuff those
digits into ${EXTEN}, therefore you need an exten = _XXX,1,Whatever if
you are expecting 3 digits.
Until recently we had DID service from our telco on an EM Wink
channelized voice T-1. The above is what we did.
David
Michael Winstead wrote:
A PRI connection is required to pass DNIS digits. Just a 24 channel wink
start T-1 with no D channel will not pass DNIS.
That would depend on how you define DNIS. Dialed Number Identification
Service. Most of the time the DNIS is the same as EXTEN, since the
dialed
C F wrote:
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog interface like FXO
the line has no way of knowing you just hung up and will continue to
ring, which asterisk will see as a new call. in my experience even
when using
Gary H. Thompson wrote:
Hi,
Most of my experience in this field have been with
Borland products, specifically Delphi. I also have been involved with
database programming, same platform as the communications.
I can't tell you anything more helpful than Michael has. I purchased a
tomb
Yes, but I would like to try a number and after to try a second one.
Any Idea how to avoid this.
On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
C F wrote:
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog
hi,
hi, i did wrote (assembler) programs for cp/m!
if your experience is more on telephony', i think you will find trixbox
easier. in one cd you will have a ready system.
if your hardware is fully recognized, great !
do not use a too old machine nor a too new one.
mind that the install will
Hi,
I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when
dialing to that PSTN line asterisk see gets the call and direct it to
the right extension but if the extension doesn't answer and the dialer
is hanging the call the extension will keep on ringing.
I'm not an expert but it seems
On Sun, Feb 18, 2007 at 10:55:20PM +0200, Itamar Lavender wrote:
Hi,
I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when
dialing to that PSTN line asterisk see gets the call and direct it to
the right extension but if the extension doesn't answer and the dialer
is hanging the
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the two values
when directing the calls to the HT488 FXS instead of asterisk it does
disconnect! that's why I believe this is has to do with asterisk in it's
busy/hangup tone detection.
Itamar.
On Sun, 2007-02-18 at 23:09 +0200, Tzafrir Cohen wrote:
Chances are that the unit does not have proper busy
Why would the card care? This would be something you'd take care of in your
dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all
the intelligence you need to do it right.
On 2/18/07, Matt [EMAIL PROTECTED] wrote:
Why would the card care? This would be something you'd take care of in
your dialplan.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
Check the value of DIALSTATUS then decide of you want to dial the 2nd
number. See [macro-std-exten] in extensions.conf for an example of
checking the value of DIALSTATUS.
The only time you might want two Dial lines in a row is if you always,
not matter what, want to dial the 2nd number.
Il
On Sun, 18 Feb 2007, Matt wrote:
Why would the card care? This would be something you'd take care of in your
dialplan.
Right, the card wouldn't care. So does Asterisk know about how to send
and receive delimited ANI and DNIS through a channelized voice T1?
--Ron
On 2/18/07, Ron Fox
Ron Fox wrote:
Arriving late to this discussion, sorry if this has already been mentioned
but DNIS and ANI can be variable length without confusion if the sender
uses the * DTMF tone as a delimiter. Thus sending *ANI*DNIS* (pronounced
Star ANI Star DNIS Star allows the receiver to identify the
Matt wrote:
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have
all the intelligence you need to do it right.
There can be many reasons not to get a PRI. Most of them have to do
with cost. Depending on the location and telco, a PRI can be MUCH more
expensive than a CT1.
On Sun, 18 Feb 2007, Matt wrote:
BTW. This seems kinda backwards. Why not just get a PRI. PRIs have all
the intelligence you need to do it right.
You may not have that option. For example, you want to split a T1 from a
legacy PBX to 12 channels to a proprietary IVR system and 12 channels
Again,
I think this would be something you'd have to do in your dial-plan. If you
want Asterisk to SEND the info. You need to code your dial plan to send it
when the call starts up.
On 2/18/07, Ron Fox [EMAIL PROTECTED] wrote:
On Sun, 18 Feb 2007, Matt wrote:
Why would the card care? This
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time
Bandit
Sent: Sunday, February 18, 2007 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Summary of Trixbox vs. custom install
snip
I also include a
I, too, have heard about that best practice of using different
channels for different AP's on the same SSID. As far as I can tell,
This is standard textbook stuff. Read Cisco press's 'Deploying License
Free Wireless Wide-Area Networks' by Jack Unger.
it's BS. I don't know who started it,
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
extension number so most of the dialplain suggestions offered so far won't
work. I did try the Wait() function as was suggested. I tried it first in an
David Ruggles wrote:
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
extension number so most of the dialplain suggestions offered so far won't
work. I did try the Wait() function as was suggested.
Eric ManxPower Wieling wrote:
David Ruggles wrote:
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a
difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
extension number so most of the dialplain suggestions offered so far
won't
work. I did try the
Stephen Bosch wrote:
Have you tried calling ATT and asking for call disconnect supervision?
I realise that this can be a thankless and tedious endeavour, but it IS
worth trying. There are almost no commercial switches that don't support
this; it's a matter of activating it for the specific
Also check out immediate=no
On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Eric ManxPower Wieling wrote:
David Ruggles wrote:
I'm sending 12345 as DNIS on a Wink Start T1. In case it makes a
difference,
I'm using a Sangoma A101 card. Asterisk sees each digit as a separate
Brian Capouch wrote:
Better luck with ATT than I had with the Monon Telephone Company.
They have a switch that's fairly new, so I called them--I'm a loyal but
tightly captive customer of the last 25 years.
Their chief technician told me, Sure, our switch is new. There's
nothing to it
Would you attach your whole zaptel.conf and
zapata.conf?
--- C F [EMAIL PROTECTED] wrote:
Also check out immediate=no
On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
Eric ManxPower Wieling wrote:
David Ruggles wrote:
I'm sending 12345 as DNIS on a Wink Start T1.
In case
Stephen Bosch wrote:
And then he hung up on me.
...wow.
This society is doomed.
Actually, it isn't so much society as the legacy telcos.
But unfortunately, they've been pretty smart about using the billions
that they've stolen from us over the years: they use a lot of it to line
the
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