similar to: implementing call center using asterisk

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on Ebay, and you > can do 48 channels per adit 600, with 2 T1 interfaces. Having

Thanks Mitul, The server spec is okay but I need information on the fxs hardware to use. Regards On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mitul at enterux.in> wrote: > Quad core Xeon with 4GB ram > On Feb 17, 2016 12:32 PM, "Goke Aruna" <goksie at gmail.com> wrote: > >> Hello all, >> Can someone recommend what hardware to use for a 1000 analogue

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070220/55700f37/attachment.htm

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

On May 16, 2016 22:15, "Telium Technical Support" <support at telium.ca> wrote: > > You don't mention a configuration generator (like Elastix/FreePBX) so I'll > assume you are using a plain old vanilla Asterisk installation. In which > case all user/endpoint information is kept in config (ini) files, and no > user/endpoint manipulation is done through the

Hi I have created a rudimentary perl script that does most of what I want but occasionally in seems that a file will not play. I see the message getting sent to Asterisk but no reply to say that it completed. In fact, the very next SAY DATE command works and everything after that but the previous message seems to be ignored. In addition, I found in one case that I had to read to STDIN

hi, I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5. I got install installed ok.. after i had disable the xpp_usb module. However, when i try to compile asterisk and having this error I will be glad for your kind response. Goksie "chan_zap.c: In function ?pri_dchannel?: chan_zap.c:9203: error: ?pri_event_setup_ack? has no member named ?call? make[1]: *** [chan_zap.o] Error

Can someone guide me on what package I can use to do least cost routing in asterisk-1.4 without going through the prepaid calling card platforms. I have tried Asterisk::LCR and LCDial without success, if more help on either too. I will be glad. I will be glad for good pointers. Thanks. Goksie

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all went okay. using sangoma a104dx on both machine. I followed the write up on http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup I have the cross over cable between them. however, wanpipe shows connected but the signaling link does not align. i have my configs for host A ##wanpipe1.conf [devices] wanpipe1 =

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure:

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0d4c2800/attachment.html>

hi all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46