Use a Call file to dial back to the PBX.
In voicemail.conf set the externnotify value to something like:
externnotify=/usr/local/sbin/mwi.pl
where the perl script creates the Call file. I set up a specific group and
dedicated a port to making these calls instead of chancing the glare with
the pb
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Ronan
>Eckelberry
>Sent: Thursday, May 05, 2005 9:09 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Put a wait in a .call file.
>
>
>No go. Now, it picks up the
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Andrew
>Kohlsmith
>Sent: Monday, April 25, 2005 8:04 AM
>Channel banks are great; the better ones (Adit600) can do far-end
>disconnect
>supervision and I think pretty much all of them do dynamic impedance
>
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Steve
>Edwards
>Sent: Tuesday, April 05, 2005 10:30 AM
>To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Can't see ANI2 (aka info digits) from PRI
>t1
>
>
>No.
There is an error in the ASTCC makefile. It places the sound files in the
wrong directory.
You can either modify the Makefile and change the line:
SOUNDSDIR=/usr/share/asterisk/sounds
to
SOUNDSDIR=/var/lib/asterisk/sounds
and then re-install. Or simply move the files from the improper direct
>my $sth = $dbh->prepare("SELECT * FROM routes WHERE " . $dbh->quote($num
>ber) . " RLIKE pattern ORDER BY LENGTH(pattern) DESC");
>
>Does it mean I just need to use:
>^61.* 100
>^6178.* 150
>^615.* 130
>^61342.* 180
Ronald,
The ASTCC sql SELECT used will return the routes entry that match
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Steven
>Critchfield
>Sent: Monday, March 07, 2005 6:08 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Question about AGI vs. FastAGI vs.
>straight C/DB developme
-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Steven
>Critchfield
>Sent: Friday, March 04, 2005 10:26 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Stutter Tone
>
>
>On Fri, 2005-03-04 at 21:10 -0600, Anton
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Ronald
>Wiplinger
>Sent: Thursday, March 03, 2005 2:47 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] ASTCC questions
>
>
>Ronald Wiplinger wrote:
>
>(Correcting
>I'm not 100% sure, but I think Fedora Core 2 uses UDEV. Look through
>the output of ps -A and see if there is a udevd running. If there is
>you're running udev and need to read README.udev which is in the zaptel
>source directory.
>
>
I'm running FC2 V2.6.10-1.14 and it is not using udev.
When y
g List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log
>
>
>Hi,
>OK, well, I've disabled SSH/HTTP already so lets hope I will have
>my system
>working!
>Best and thanks,
>Christian
>
>
>- Original Message -
>From: &
>
>
>On Thu, 2005-02-10 at 10:56 -0500, Karl H. Putz wrote:
>> I had the system setup to allow http and ssh.
>>
>> The hack came in through ssh.
>
>I doubt you where hacked via ssh. Most likely you had your password
>brute force cracked.
That is what I meant to re
mercial Discussion
>Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] scary log
>
>
>Hi,
>I've also been a little worried about the security. How did they
>connect to
>your system? Through telnet or what?
>Since I've disabled all such services.
>Best,
>Christian
>
You've likely been hacked.
I have recently had a similar incident where a hacker guessed my root
password (MY BAD) and set up an ebay password skimming site.
I noticed it when I got similar non-deliverable email messages.
Obviously, first change your password and then look at the /var/www/html
d
remember to use the ^ to indicate matching at the beginning of the number.
i.e ^01144 should be all you need to match any international call going to
country code 44.
Karl Putz
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Jason
>Kawakami
>Sent: Wedne
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Mike Nugent
>Sent: Tuesday, February 08, 2005 5:31 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] astcc with multiple access
>
>
>
>
>I'm looking at astcc and it seems that setting up a scr
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of thieumS
>Sent: Tuesday, February 08, 2005 10:09 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] ASTCC simultenous calls per card
>
>
>Hi guys,
>do you know if it'
Trevor,
The problem is a "division by zero" issue in the astcc.agi @ line 538 (I
have made a few mods so my line #'s may not be exactly the same). The line
reads
:
$maxmins = int(($credit - $adjconn) / $adjcost);
you may want to change the script to something like:
if ($adjcost < 1) {
>The SQL statement in astcc returns all the matched patterns with the
>longest, most specific match first and uses only that first match in its
>processing. So you could also use the pattern: '.' to match any dialed
>number not already matched as a default BUT BE SURE to set that cost high
>enoug
-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
>Sent: Friday, February 04, 2005 4:50 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] ASTCC Apllication
>
>
>Hello,
>I have some problem using ASTCC applic
> El Jue 03 Feb 2005 14:02, Steve Totaro escribió:
> > does anyone know how to change the timeout on digit entry in
> astcc. if you
> > call the app and start entering a pin, you have about 2 seconds to enter
> > the next number or you get timed out. i cannot find any info
> on this from
> > the
The current astcc Makefile puts the sound files into the wrong directory.
It uses /usr/share/asterisk/sounds but it should be
/var/lib/asterisk/sounds.
Karl Putz
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Daniel Eboa
> Sent: Saturday, January 29,
Sangoma is planning to release a multi-port (T1/E1) card later this year
with DSP resources available.
Karl Putz
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Michael
> Baird
> Sent: Thursday, January 20, 2005 2:42 PM
> To: Asterisk Users Mailing Li
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Michael K.
> Rodriguez User
> Sent: Thursday, January 20, 2005 2:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] PRI info digits question
>
>
> Does anyone kn
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Tony
> Mountifield
> Sent: Friday, January 14, 2005 12:56 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4
> server
>
>
> In article <[EMAIL PR
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> Totaro
> Sent: Wednesday, January 12, 2005 12:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] ASTCC configuration problem
>
>
>
>
> > > > If not
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> Totaro
> Sent: Wednesday, January 12, 2005 10:30 AM
>
> If nothing else, my efforts are documented for anyone else in the
> same boat.
> It seems that you can debug agi by typing agi debug at the *
Steve,
What version of MySQL are you running? I upgraded to 4.1.8 and ran into the
problem below. I initially tested with the user root and the default blank
password and was OK. But when I changed over to a new user with a password,
I noticed an error message in the httpd logs:
Client does no
> CREATE TABLE `cdrs` (
>
> `cardnum` char(40) default NULL,
>
> `callerid` char(80) default NULL,
>
> `callednum` char(80) default NULL,
>
> `trunk` char(40) default NULL,
>
> `disposition` char(20) default NULL,
>
> `billseconds` int(11) default NULL,
>
> `billcost` int(11) default NULL
> ) TYPE=
There is a field missing in the admin.cgi CREATE for cdrs.
add: callstart CHAR(24) to the cdrs table
There is a patch to fix the cgi at
http://bugs.digium.com/bug_view_page.php?bug_id=0002796
It just hasn't made it through to CVS yet.
Karl Putz
> -Original Message-
> From: [EMAIL PR
I have been having this exact problem with a Tatung dual EMT-64 server as
well.
I have been trying to get a TE410P running and all looks great, driver
loads, runs ztcfg OK, etc. but no interrupts are ever processed.
One additional piece of info that I have not seen in this thread is that I
am abl
> > I really am at my wits end about this one. Some people report this
> > card and server working fine while others (like myself) can't get it
> > going no matter what. I have been told by the Digium distributor in
> > our country that this card simply "not compatible with some
> > motherboards".
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of
> [EMAIL PROTECTED]
> Sent: Monday, January 03, 2005 1:35 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
>
>
> Hi,
>
> Has anyone had succ
The AGI looks in
/var/lib/asterisk/sounds for its sound files with names not fully
qualified.
This looks like a bug in
the ASTCC makefile or the astcc.agi should refer to the soundfiles in the
/usr/share/asterisk/sounds directory explicitly.
Karl
Putz
-Original Message-From
gt;
>Darren Wiebe
>[EMAIL PROTECTED]
>>Karl H. Putz wrote:
>> I am looking for the most stable version of Asterisk to use with ASTCC
>> for a production environment.
>>
>> It does not appear that any of the Stable versions will be suitable
>> since they do not sup
I am looking for the most stable version of
Asterisk to use with ASTCC for a production environment.
It does not appear that any of the Stable
versions will be suitable since they do not support US PRI ANI Info digit
collection and hence could not apply surcharges for payphone use,
etc.
Gunnar,
Add the voicemail options you want to use for that mailbox to the "users"
table as text with multiple option=value pairs separated by a '|' (pipe)
character just as it was in the voicemail.conf file.
I believe the envelope=yes option is what you want to set.
Karl Putz
Forte Communicatio
added.
Matthew
- Original Message -
From: "Karl H. Putz" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, October 18, 2004 10:00 AM
Subject: [Asterisk-Users] New Realtime config and MWI
>
Does the new Realtime config in the CVS head support setting and clearing
MWI for sip clients?
Thanks,
Karl Putz
Forte Communications
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