search for: ccmanager

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