We've been happy with the polycom IP 7000.
Darren Wiebe
On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote:
Hi Faisal,
Thanks for reply but I want hardware wase VoIP device. If know please
gussed me. From google I fould the list of below devices but I am not sure
=Main_Page if you want to do
something similar. Why try to make Asterisk into something it's not
intended to be? Just use your firewall for what it's good at.
--
Darren Wiebe
On 7/23/11 11:38 AM, CDR wrote:
I beg to differ. Digium is hiding from the real world and somebody is
going take the software
interested the information is available here:
http://fail2ban.aleph-com.net/fail2ban_sharing If you're interested in
the server code just drop me an email.
Darren Wiebe
dar...@aleph-com.net
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We recently completed a project using products from here:
http://www.controlbyweb.com/webrelay/ They were easy to setup and can
be controlled in a variety of fashions included http queries.
Darren Wiebe
On 18/10/2010 8:34 AM, Marco Signorini wrote:
Hi
Did you looked at Arduino + Ethernet
when you're done. You can
also disconnect calls from the asterisk cli using the soft hangup command.
Darren Wiebe
dar...@aleph-com.net
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New to Asterisk
) or M0n0wall. I've had good luck
with both of those.
Darren Wiebe
dar...@aleph-com.net
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk
.
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Darren Wiebe
dar...@aleph-com.net
Aleph Communications
www.aleph-com.net
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purchased a few copies of it is that
I need to have several different sip and iax2 connections for testing
purposes.
--
Darren Wiebe
dar...@aleph-com.net
Aleph Communications
www.aleph-com.net
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My thoughts were similar. Availability has not been a problem for us on
the WRT54GL boxes. We're pulling them out of our wholesaler all the
time without any problems.
Darren Wiebe
dar...@aleph-com.net
Jeff LaCoursiere wrote:
And why not DD-WRT, which runs on many more platforms including
Just restarting it won't do anything. You could use the following
command to find any files over 200mb on the system. Be careful about
blindly deleting stuff though
*find / -type f -size +200M
Darren Wiebe
dar...@aleph-com.net
*
David @ULC wrote:
I have 320 GB SATA HDD.
When I
be happy to try it again to see if I've become a male yet. :)
Darren Wiebe
dar...@aleph-com.net
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Pretty cool. I'm almost offended though as I'm not usually guessed as a
female of the species. :)
Darren Wiebe
dar...@aleph-com.net
Asterisk Asterisk wrote:
Steve,
Tried to test and got call could not be completed as dialed.
Were you able to connect? If not, please try again. Call volume
We've done this with good results. You can also get one that flashes a
bright light for not a lot of money.
Darren Wiebe
dar...@aleph-com.net
Steve Gladden wrote:
If you wanna go low tech. down dirty you could also go with a conventional
POTS phone line 'loud ringer' device and simply hook
do not need to link to Asterisk, etc.
Darren Wiebe
[EMAIL PROTECTED]
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For simple paging the bogen tamb works very well. Just hook it up to an
fxs port and you're good to go.
Darren Wiebe
[EMAIL PROTECTED]
Jonathan Disher wrote:
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr.
Darren Wiebe
[EMAIL PROTECTED]
emist wrote:
Hello,
does anyone know of a good calling card solution for asterisk that is
able to do lcr?
Does astcc do this? I've been searching around and I can find some lcr
modules
Just FYI, I wrote an application that tracks the status of SIP or IAX2
extensions by listening to the AMI. It was for use by callshops but
would probably require minimal change to work for you. It's currently
part of the ASTPP source code.
Darren Wiebe
[EMAIL PROTECTED]
Atis Lezdins wrote
If you're willing to cc me a copy I'll be in your debt.
Thanks,
Darren Wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED]
wrote:
Steve Totaro wrote:
On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED
Am I correct in thinking that one application of this would be
monitoring what you have left for funds with a prepaid vendor?
Darren Wiebe
[EMAIL PROTECTED]
Brian J. Murrell wrote:
On Wed, 2008-04-23 at 09:38 -0700, Nhadie Ramos wrote:
Hi, sorry to confused you with my question
Ok, I'm not aware of this feature in astcc and I can't speak for astbill
or a2billing. I do know that I coded it into astpp and it's called
vendor rating in there. It works but it's not used a lot at present.
Darren Wiebe
[EMAIL PROTECTED]
Nhadie Ramos wrote:
hi sir,
yes that would
an accountcode set. You do not need to use it to manage
your dids and extensions, etc.
Darren Wiebe
[EMAIL PROTECTED]
Vicky wrote:
I am also searching one for post-paid billing .. but most like astpp
wants to eat whole system themselves managing extensions and all . I
need a type of solution
Ok, cool. If you run into problems, please post at forums.astpp.org on
the the astpp mailling list.
Good luck,
Darren Wiebe
[EMAIL PROTECTED]
Vicky wrote:
I will definitely give it a try again to astpp then . I actually saw
its online demo and was bit confused
. I thought its managing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a script to do this found here:
http://www.astpp.org/index.php?n=Misc.AutoDialOut
Darren Wiebe
[EMAIL PROTECTED]
Tom Engleward wrote:
--- Angus Comber [EMAIL PROTECTED] wrote:
I have been asked by a client to process a list of telephone
an issue but on prepaid it
still is.
Darren Wiebe
[EMAIL PROTECTED]
Jon Farmer wrote:
JP Carballo wrote:
Yes, certainly, through deadagi.
I just have one question though, why reinvent the wheel?
There are prepaid systems that work with asterisk.
I have yet to find a prepaid system
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
chawki hammoud wrote:
With a few fairly minor programming revisions to the script this would
be possible. At present, ASTCC does not support that though.
Darren Wiebe
[EMAIL PROTECTED]
Hi users:
astcc script exits when dialing an uncomplete
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
Checkout options H and h.
Darren Wiebe
[EMAIL PROTECTED]
Obelix wrote:
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application
= _1XX,100,Congestion; No Route Available
# exten = _1XX,101,Hangup
Some of the group counts are for outgoing trunks. It's just the first
one that you need.
- --
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Here's how I do it. I have the phones tagged to accountcodes and I use
the channel counting features of asterisk to limit an accountcode to X
number of simultaneous calls.
Darren Wiebe
[EMAIL PROTECTED]
Bryan Mahin wrote:
Lol.. To an extent I
of effort. Unless, of
course, you did all the coding for this yourselves. If there is a
major disagreement or problem with it I can see branching other than
that it seems unfortunate.
Just my $0.02CDN.
Darren Wiebe
[EMAIL PROTECTED]
Mindaugas Kezys wrote:
!-- /* Style Definitions
/index.php?n=ASTERISK.Code
Eric Wieling, if you want the changes I made you're more than welcome
to them and the sound files also. :-)
Darren Wiebe
[EMAIL PROTECTED]
Cosmin Prund wrote:
Hello everyone.
This is an other question from a relatively newbie.
I'd like to provide auto callback ability
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, you can do that using ASTPP (www.astpp.org). You can bill DIDs
per month and map them to the appropriate ATA device using the gui.
Darren Wiebe
[EMAIL PROTECTED]
Bernard Cresencia - CrossNet International wrote:
Hi all,
Aside from
.
Darren Wiebe
[EMAIL PROTECTED]
Chris Mason (Lists) wrote:
I am copying the Master.csv file to another server and importing to
mysql. I am looking for a simple billing application that will
produce a bill for a give account code for a give period, based on
a rate table. Is this available
payments... ASTPP does not have the capability at present
to process credit cards or Paypal. This is another which I have been
using oscommerce for. You can sign up for a voip account and refill
your account using our oscommerce plugin.
If I missed something let me know.
Darren Wiebe
[EMAIL PROTECTED
but it does keep the call on
the local host.
Darren Wiebe
[EMAIL PROTECTED]
Jeremy wrote:
!-- /* Style Definitions */ p.MsoNormal, li.MsoNormal,
div.MsoNormal {margin:0in; margin-bottom:.0001pt; font-size:12.0pt;
font-family:Times New Roman;} a:link, span.MsoHyperlink
{color:blue; text
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a perl app that listens for hangups and then grabs the call out
of the database using the uniqueid. Maybe not the neatest way but it
works well.
Darren Wiebe
[EMAIL PROTECTED]
Mark Ackroyd wrote:
The reason for it being 0 is because
of Asterisk. Any comments/suggestion?
Darren Wiebe
[EMAIL PROTECTED]
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEHjLG4DADnh+tnOQRArn+AJ0dx9fncjX77QVtP0VzCXqa2i0BXwCdFv1v
0UQ9s6cloDFZJwIiBWJe/Hg=
=fi8U
my ( $var1, $var2, $var3 ) =
@ARGV;
and so on and so forth.
Good Luck
Darren Wiebe
[EMAIL PROTECTED]
Paul Hales wrote:
Thanks for this example - it has really got me started!
Short question - how can I put a variable into my perl script?
I imagine it's something like
exten = 780,1,AGI
://lists.digium.com/mailman/listinfo/asterisk-users
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Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Asterisk-Users
I think that the feature you're looking for is called pricelists in
ASTPP but I could misunderstand what you want. Feel free to post the
question either on the astpp-users mailing list or the astpp forum.
Visit www.astpp.org for more info.
Darren Wiebe
[EMAIL PROTECTED]
Pavel Jezek wrote
in place already.
/blatant plug ends/
Hope this helps
Darren Wiebe
[EMAIL PROTECTED]
Damon Estep wrote:
Does anyone have a mysql query that will compare a number from the
asterisk cdr to a table of international country+city codes to
determine the closest match?
The two fields are;
1
Hours of struggling later, I have found the problem. Here is the
correct format for those outgoing calls.
SIP/[EMAIL PROTECTED]||L(54081429:6:3)|Hj
I'll try to get a patch done up one of these days.
Darren Wiebe
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On 00:30, Mon 06 Feb 06
Well, I'm not real sure on whether I like the idea or not bug
Anyway, here is an app that I wrote for something similar to this. It
was for notifying customers of events,etc.
http://www.astpp.org/index.php?n=Misc.AutoDialOut
Darren Wiebe
[EMAIL PROTECTED]
Ron Senykoff wrote:
Hi
It's part of ASTPP. It is in astpp -head ready for testing.
Darren Wiebe
[EMAIL PROTECTED]
Sam Tam wrote:
When will it be ready ?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: Saturday, February 11, 2006 9:38 AM
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[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph
This is hopefully on topic. I'd like thoughts on this. I'm looking at
doing some dialplan work which would grab the sip devices IP number. If
that ip number is in an allowed list, the call would be allowed to go
through otherwise congestion would be passed. Any thoughts?
Darren Wiebe
This doesn't really belong on the asterisk-users list. ASTPP has it's
own mailing list. This can be found @ www.astpp.org. I, or someone
else will be happy to help you either there or on the forums. On your
1st post please mention what version of ASTPP you are using.
Thanks,
Darren
but I am behind. Check
out the astpp demo. www.astpp.org
As for * on ser, you may want to visit :
http://www.voip-info.org/wiki-SIP+Express+Router
Good Luck,
--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
I'll make progress on the weekend. :-)
Darren Wiebe
[EMAIL PROTECTED]
JP Carballo wrote:
Darren Wiebe wrote:
JP Carballo wrote:
Ronald Ramos wrote:
Hi All,
Any solution on how I can implement prepaid billing on asterisk?
But not the calling card type, just a simple Custome rwill buy
credit
Have you settled on a calling card application yet? There are a host of
different options. I, of course, recommend astpp. :-) The wiki will
have much of the info you will need.
Darren Wiebe
[EMAIL PROTECTED]
Dirgan Putra wrote:
Iam new in asterisk user, can helpme to install asterisk
Cool! I just tool a look at it looks like you did a great job!!
Darren Wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
Sorry if this is slightly off topic but it does pertain to Asterisk
Users as well as the biz list. Also, sorry if it is a double post but
the first one never made
http://www.astpp.org/index.php?n=Misc.AutoDialOut
I put together what I have on that site.
Darren wiebe
[EMAIL PROTECTED]
Steve Totaro wrote:
Darren,
I am interested in your project. Let me know if I can help you test.
Thanks,
Steve
-Original Message-
From: Wiley Siler
://lists.digium.com/mailman/listinfo/asterisk-users
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Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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that Darren, I was looking at your scripts
today. I will evaluate it some more.
On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote:
I have written an agi script that I use for that. Then I can just have
a list of dids and extensions in a db.
Tom Vile wrote:
Would like some advice
Local/[EMAIL PROTECTED]
Try putting the context on. I do this all the time in callfiles.
Darren Wiebe
[EMAIL PROTECTED]
Matt wrote:
Hi,
What do I have to do to get local\number to work in a context?
It works from my [from-internal]... however from subcontexts it does not work:
Jan 6 15
I'll try to finish this up tonight and post back once I'm done.
Darren Wiebe
[EMAIL PROTECTED]
Wiley Siler wrote:
If this or any other example is available, I would be most thankful to
have it.
I got the go ahead on this project to day so now I have to start seeing
how to do this.
Thanks
I'm supposed to have a mostly canned script that will do this done
already. It will pull the list of people to call out of a db and play
them the file specified in the db table. Contact me offlist if you're
interested. It will be done real soon but I'm not done testing yet.
Darren Wiebe
There is a web interface. It's pretty basic but you can find a demo
here: http://dc.maxnet.ru/cpdemo/ I know the guy that owns it.
Contact me if you're interested. It's payware.
Darren Wiebe
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hello,
Is there a GUI to manage the users
That is because they switched over to svn I belive.
Darren Wiebe
Colin Anderson wrote:
cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel'
Anyone else seen this?
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You know, that's right. I thought so too. I've been entirely
unsuccessful getting cvs downloads but that could just be my luck.
Merry Christmas Everyone,
Darren Wiebe
[EMAIL PROTECTED]
Colin Anderson wrote:
I thought they were going to run CVS concurrently for a while??
-Original
http://www.voip-info.org/wiki/view/Asterisk+bounty
Darren Wiebe
[EMAIL PROTECTED]
Douglas Garstang wrote:
Digium needs people like me, if they read this list that is. They sure don't
seem to be able to make real-world functionality decisions on their own.
-Original Message-
From
Everybody is entitled to their own opinion. I believe Kevin Fleming
indicated that the Digium todo list was flexible if enough $$$ of
funding were involved. Maybe that would interest you more.
Darren Wiebe
[EMAIL PROTECTED]
Douglas Garstang wrote:
I don't think the bounties are worth
of options.
1. Hire a consultant or programmer to fix it for you.
2. Try to hire Digium to fix it for you.
3. Find another application that works for you.
Darren Wiebe
[EMAIL PROTECTED]
Douglas Garstang wrote:
I don't know what the rules are for this list, but it wouldn't be much
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[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
this with prices but the calling card stuff is
only in cvs yet.
Darren Wiebe
[EMAIL PROTECTED]
jonny hashem wrote:
Hi list:
I need to create a routes list to specific card number
wih different prices than the initial routes list
,because markup donot achieve my purpose and markup
use
I think you have to run the update_database function. That code was
written over a year ago and has not been touched since that I'm aware
of. I suspect the Friends support should be moved over to realtime but.
Darren Wiebe
[EMAIL PROTECTED]
Insider KT wrote:
Hi. I am having some
Try it out. It looks to me like it would work but I've been wrong
often. :-)
Darren Wiebe
[EMAIL PROTECTED] wrote:
List ... Darren,
In order to use a provider with unusual prefix 00
i.e. 001NXXNXX and providing failover to other providers with
the usual 1NXXNXX, decided to:
1
If it works fine, I can't think of any collateral damage. All that
affects is the way the string is put together.
Darren
[EMAIL PROTECTED] wrote:
Thanks.
It works fine. I was just curious about
any collateral damages.
Thanks again,
benchev
On Monday 12 December 2005 16:42, Darren Wiebe
You have to do that from the dialplan. I have a script that looks up
the DID in a database and sets the accountcode. It does some other
stuff also but that could easily be cut out. It's part of ASTPP. Drop
me a line if you need a copy.
Darren Wiebe
Matt wrote:
Hrmm that works except
have. You would want to replace the meetme stuff with
whatever you want the other end to connect to.
Darren
wassim darwish wrote:
Hi:
Once i have seen the post of Darren Wiebe of
suggestion of a callback configuration in
extensions.conf and it was like this:
[callback]
exten =
_.,1,AGI
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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is the context to throw the call into when it's connected
enhanced-outgoing is where I send the outgoing calls through. I use the
local channel.
Make sense?
Darren Wiebe
[EMAIL PROTECTED]
chawki hammoud wrote:
Hi:
i tried to lauch the callback.agi script and astcc.agi
script together but i failed to do
want or visit www.aleph-com.net/astpp
Darren Wiebe
[EMAIL PROTECTED]
Chris Bagnall wrote:
Good morning all,
I'm trying to find an application that'll do really lightweight billing for
Asterisk CDRs.
On our asterisk servers deployed at people's offices, we have CDRs being
logged to PostgreSQL
to figure it out by the cron job and
reset it from there.
This would be fairly easy. You would need a script that ran the
appropriate sql command when called.
Is a solution available?
Yes, it is.
Good Luck Ronald, I haven't talked to you for a long time. :-)
Darren Wiebe
[EMAIL PROTECTED
onto the end of your dialcommand. You can look at the
code for ASTCC in the asterisk cvs or look at astpp-callingcard.agi in
the cvs code available @ www.aleph-com.net/astpp
Darren Wiebe
[EMAIL PROTECTED]
Innocent Evil wrote:
Comeo'n AGI guys..
Please say something.
Hi,
Using
I just upgraded to 1.2 and that fixed the problem for me.
Darren Wiebe
[EMAIL PROTECTED]
Abdock wrote:
Hello,
Getting this error and the audio is too low,
file.c:550 ast_readaudio_callback: Failed to write frame
How to get correct
Use a line like this in your dialplan. I'll post a sample out of mine.
exten = _NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1780${EXTEN}
That line is setup so any 7 digit numbers will be marked as belonging to
1-780.
Darren Wiebe
[EMAIL PROTECTED]
Jason Brashear wrote:
How do you set it up so
]
All outgoing calls will be placed through the Local channel in context
enhanced-outgoing.
Hope this helps
Darren Wiebe
[EMAIL PROTECTED]
Musaluke AK wrote:
Darren,
An example how to call that callback.agi script? The script iself does
not have usage info.
Thanks
Anthony
Darren Wiebe
-com.net/astpp. Somewhere there. It is way more complicated
than you need but you can cut out all the user interaction stuff.
Darren Wiebe
[EMAIL PROTECTED]
Abdul Lateef wrote:
Hi friends,
I am new in asterisk, i came for CallBack purpose, i
read from Voip-info.org aboue callback with asterisk
I only have the answer to your last question. From my experience, I
would go for arbitrary barf. I don't think you are supposed to get
anything if there is not a caller id passed.
Darren
Dave Grey wrote:
Well, I am batting close to zero where responses to my questions are
concerned, but
We don't have a complete package quite yet. I think we have most of
what you will need but we do not have support at present yet to accept
customers payments. We can do that easily via 3rd party sofware but we
can't do it ourselves yet. Anyway, www.aleph-com.net/astpp is the link.
Darren
I've been thinking of using yate
http://yate.null.ro/pmwiki/index.php/Main/H323ToSIPSignallingProxy to do
this. Any thoughts or experiences?
Darren Wiebe
[EMAIL PROTECTED]
Rob Lith wrote:
Altus
It's in the transcoding -
http://www.voip-info.org/wiki-Asterisk+dimensioning has some notes
I don't know if astbill supports this or not. ASTPP does supports it
though. www.aleph-com.net/astpp You would set admin 1 and admin2 up
as resellers.
Darren Wiebe
[EMAIL PROTECTED]
Kanishka Somaratne wrote:
Hi
I am looking for a asterisk billing system with a reseller module
What channel are you using to place the calls from ASTCC and what
version of asterisk are you using? The get_variable and set_variable
perl commands are not working in -HEAD due to stuff being deprecated.
Darren Wiebe
[EMAIL PROTECTED]
maka wrote:
Hello list,
I just came into a strange
with the problem anyway..
Wasn't this fixed a while ago? I had a patch that I thought had been
accepted..
Darren Wiebe
[EMAIL PROTECTED]
On 10/19/05, *Darren Wiebe* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
What channel are you using to place the calls from ASTCC and what
This will actually be easy to fix. I'll post a patch along with
someother stuff shortly.
Darren
Darren Wiebe wrote:
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL
We had this problem a few months ago but they resolved it for us. I
really don't remember more than that.
Darren Wiebe
[EMAIL PROTECTED]
Tom Vile wrote:
I have been battling this problem for 2 months with no resolution as
of yet with TelaSIP. I am told that it is a provider problem(Level 3
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTECTED]
Eric Lyons wrote:
Looking at the code, it would appear that the 'callstart' column of
the cdrs table should
The way I do it is to make a list of internal extensions and set those
to no charge. They get billed at no charge that way and it works fine.
/Plug Starts/ This is done using ASTPP www.aleph-com.net/astpp/ /Plug Ends/
Darren Wiebe
[EMAIL PROTECTED]
Chris Bagnall wrote:
Hi all,
I have
Thanks. I have a question for the mailing list in general. Where
should the card get marked as in use? Should it be as soon as you enter
the number or should it be when it dials? I don't know for sure.
Darren Wiebe
[EMAIL PROTECTED]
Michael K. Rodriguez wrote:
This is my debug
to notify the
caller at the earliest. This shouldn't happen of course.
Darren Wiebe wrote:
Thanks. I have a question for the mailing list in general. Where
should the card get marked as in use? Should it be as soon as you
enter the number or should it be when it dials? I don't know for sure
Edit astcc.agi and stick these lines in before sub load_config.
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
How do you you apply the patch?
-Scott
- Original Message - From: Nicolás
Can you please post the output with debug agi on ?
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
I download and installed ASTCC over the weekend and I am having an
issue where the INUSE flag will not get set back to 0 if the user
drops a call while the balance is being played. All other
I fought with this one for hours last night. I have to get it yet but
I'm not sure what the problem is. The permissions are all fine.
Any comments anyone?
Darren Wiebe
[EMAIL PROTECTED]
Scott Wolfe wrote:
I just installed the CVS 9-22 and am trying to get ASTCC up and
running. I was able
Okay, after spending 12 hours on it I checked the thing that has bit me
before. Turn SElinux off.
OUCH!! :-)
Darren Wiebe
[EMAIL PROTECTED]
Darren Wiebe wrote:
I fought with this one for hours last night. I have to get it yet but
I'm not sure what the problem is. The permissions are all
Could you post an example of you cdr output. The ASTPP question would
be better put on astpp-users. Visit
http://aleph.aleph-com.net/mailman/listinfo/astpp-users to subscribe.
Darren Wiebe
[EMAIL PROTECTED]
FaberK wrote:
Hi to All,
I've an Asterisk CVS Head working with Mysql.
My problem
I will look into this and post back what I find.
Darren Wiebe
[EMAIL PROTECTED]
Ricardo Poppi wrote:
Yes Darren. The problem is the same using Zap or SIP. I had no
oportunity to verify that using IAX or E1/T1.
Rgds, Ricardo Poppi.
___
--Bandwidth
I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by
[EMAIL PROTECTED] on a i686 running Linux.
I just spent some time in testing this. I tested the local and IAX2
trunks. Both worked flawlessly.
Any comments?
Darren Wiebe
[EMAIL PROTECTED
I just tested this on Asterisk CVS-v1-0-09/22/05-22:23:34 built by
[EMAIL PROTECTED] on a i686 running Linux
and it still works perfectly.
Darren Wiebe
[EMAIL PROTECTED]
Darren Wiebe wrote:
I'm running Asterisk CVS-v1-0-06/06/05-17:29:02 built by
[EMAIL PROTECTED] on a i686 running Linux
Have you done any testing to see if it made any difference what type of
trunk was being used?
Darren Wiebe
[EMAIL PROTECTED]
Ricardo Poppi wrote:
Hi all.
I´ve found a kind of solution (if we can call it this way...) and Im
reporting it here to help save some lives.
Editing into astcc.cgi I
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