Displaying 20 results from an estimated 21 matches for "ccmanager".
2007 Mar 14
3
Call center manager for Asterisk (Release 0.3)
Hi
i just want to let you know that is available a new release of ccmanager.
I've added the possibility to import queue_log information in a mysql
database and to generate reports using this information.
The software is in a beta state and provides this functionality:
- users management
- call generation (making a GET or POST request on a certain URL)
- queue manage...
2008 Jun 14
1
play sound on a specific channel
Hi to all
can i play a sound or a dtmf tone on a specific channel using AMI?
Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2007 Oct 27
0
Call center manager for Asterisk (Release 0.5)
CCMANAGER 0.5 released!!
NOTE:
this is a previous alpha release, maybe there is some customization to
do on the settings files,
i can't write a clear and complete howto at the moment
I don't have released upgrades in the last months but the project is still alive
i'm too busy at the moment, i&...
2007 May 07
2
h323 problem with asterisk 1.2.18
...grammi/asterisk_1.2.18/asterisk-1.2.18/channels/h323#
make opt
make: *** No rule to make target `opt'. Stop.
why?
where am i wrong? i've also tried the last version of pwlib and
openh323, but without fixing the problem
thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Jun 29
4
asterisk call unique id in dialplan
...lplan?
For example:
exten => 203,1,Answer
exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id})
exten => 203,3,Dial(SIP/203)
Can i do something similar that?
How can i retrieve the unique_id generated?
thanks.
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Aug 03
2
partial ChanSpy
Hi
is it possible to spy (not record, spy) partially on a channel?
for exaple, i'd like to listen only the input or the output voice.
is it possible?
thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Sep 05
1
Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
...re that implements SIP protocol.
Can you help me to guess where is the problem?
if i try to create a call from SJphone 2 SJphone all works fine.
Is possible that exists a problem in asterisk ?
where ? how can i find it ?
thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2008 Jul 17
1
OpenH323 and ptlib version for asterisk 1.4.21.1
Hi what version of openh323 and pwlib are suggested for asterisk
1.4.21.1.? Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2007 Apr 21
3
FAX on PRI and TE205P
Hi
i have a PRI connected to a TE205P.
Actually, can i send and receive FAX through Asterisk using stable solutions?
Or shall i connect an ATA to Asterisk and then a modem with Hylafax?
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 May 10
2
force outgoinc callerid
...xt?
for example:
[outgoing_context_one]
;force callerid to 12345
exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN})
[outgoing_context_two]
;force callerid to 22222
exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN})
Can i do that?
thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Jun 19
2
PhpAgi call generation
...riginate("Zap/g1/1","number","default","1");
/*
play message...
*/
} else {
die("error\n");
}
?>
But it doesn't work.
Is it possible to create a program like this?
thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Sep 13
0
[phpAGI] generate a call from a SIP device to Asterisk
...e at ip_of_device,2000,"default","1");
.....
And on the extension 2000 in the context "default"
exten => 2000,1,ChanSpy(|g(100))
exten => 2000,2,Hangup
Is it correct ?
or shall i do something else?
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Oct 03
0
multiple iax users on the same host
...lafax are both on the same lan).
Can i setup on the same host (Hylafax) multiple iax accounts ? (each
account is used by a iaxmodem instance).
The account can be on the same port or should i change the port for
each iax account?
Thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
2007 Nov 20
1
store 2 separate records in cdr when a call is transferd
...com/pipermail/asterisk-dev/2007-May/027666.html
I just want to know if there are some upgrades... on 1.4 or 1.2.
I'd like to store two records in the CDR instead of one, when a call
is transferd.
Is it possibile now?
Thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2007 Dec 11
0
new Asterisk installation with openvox 1.2 or 1.4?
...B800P
1 OpenVox A800P01
4 OpenVox FXS-100 FXS100
4 OctWare SoftEcho SOFTECHO
Do you suggest 1.2 or 1.4 branch?
Is now 1.4 stable ?
I've tried 1.4 the last year but i've experienced many problems with
misdn drivers.
Thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2008 Jan 08
2
disable call waiting by default
...ard.
I want to disable by default the call waiting sound.
I know that dialing *70 before to call the call waiting is disabled
until the next call, but isn't there a setting or a dialplan command
to set up this automatically?
Thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2008 Jun 04
0
Patch for app_asr.c: DTMF instead of goto
...to the 200,300 or 400 extension.
With the modified app_asr you will hear (and Asterisk can detects, via
AGI or dialplan) 200,300,400 DTMF tones.
You can find more information here.
http://www.kumbe.it/pagine/dettaglio/34/206.html
Bye
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2008 Jul 09
0
disable DTMF on a particular channel
Hi to all
is it possibile (via AMI or dialplan) to disable the DTMF tone on a
particular channel?
Thanks in advance
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser
2007 Feb 22
3
queue information into db
Hi
the new asterisk 1.4 supports to store queue log information directly
into a database? (like CDR) ?
thanks
2007 Dec 23
3
OpenVox A800P01 and ZT_CHANCONFIG failed
...Kewlstart (Default) (Slaves: 05)
> >>
> >> 5 channels to configure.
> >>
> >> ZT_CHANCONFIG failed on channel 1: No such device or address (6)
> >>
Can you help me to guess the problem?
thanks
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser