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Open source easy-to-customize call center solution?

不问归期 提交于 2019-12-06 16:26:04
问题 Note: A have asked this question at the Superuser community first, but since it involves a lot of software-building-related topics I decided to move it here. I need to provide a simple call-center solution to a small local business. They have a local 6-digit land line number. They accept calls locally, i.e. nobody calls them from another cities. The problem is that they started to lose customers, because with a certain amount of calls every day it has become impossible to reach the local

Genesys Platform : Get Call Details From Sip Server

十年热恋 提交于 2019-12-06 05:08:38
I want to get Call Details from Genesys Platform SIP Server. And Genesys Platform has Platform SDK for .NET . Anybod has a SIMPLE sample code which shows how to get call details using Platform SDK for .NET [ C# ] from SIP Server? Extra Notes: Call Details : especially i wanted to get AgentId for a given call and From Sip Server : I am not sure if Sip Server is the best candiate to take call details. So open to other suggestions/ alternatives You can build a class that monitor DN actions. Also you watch specific DN or all DN depending what you had to done. If its all about the call, this is the

saving data into custom cdr field

自古美人都是妖i 提交于 2019-12-05 16:13:06
I created custom field "rec_name" id table "cdr", database "asteriskcdrdb". In this field I want to store recording name. I know I should do it by adding this line in one of .conf files, but where? exten => s,1,set(CDR(rec_name)=${CALLFILENAME}) I can do it by sql statement, too, but I don't know where is the file that saves details of call into database. For mysql you need add into /etc/asterisk/cdr_mysql.conf [aliases] rec_name=rec_name If you HAVE aliases section, just add to it rec_name=rec_name 来源: https://stackoverflow.com/questions/10499261/saving-data-into-custom-cdr-field

getting the group name to the according pri port in asterisk

匆匆过客 提交于 2019-12-05 01:39:07
I am using sagoma 8 port card My chan_dahdi.conf to configure the ports are ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit ;autogenrated on 2015-06-12 ;Dahdi Channels Configurations ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no

Open source easy-to-customize call center solution?

假如想象 提交于 2019-12-04 22:06:46
Note: A have asked this question at the Superuser community first , but since it involves a lot of software-building-related topics I decided to move it here. I need to provide a simple call-center solution to a small local business. They have a local 6-digit land line number. They accept calls locally, i.e. nobody calls them from another cities. The problem is that they started to lose customers, because with a certain amount of calls every day it has become impossible to reach the local phone number (it is always busy). So, there is a call center solution needed. I assume that I can somehow

How to make asterisk server automatically response to SIP call?

∥☆過路亽.° 提交于 2019-11-29 17:36:08
My objective: I want to use softphone(3CX phone) register with asterisk server, and make call to the server and asterisk act as a server to automatically response something, like play a song. How i did: I installed asteriskNow using virtualbox, and registered the softphone by setting exntension for my SIP device (extension 333). And i write a dialplan in etc/asterisk/extensions.conf . The dialplan is : [incoming] exten =>s,1,Answer() exten =>s,n,Playback(dir-intro-oper) exten =>s,n,Hangup() I want any incoming call to server, the server will automatically answer, and play a pre-defined voice

Asterisk AMI - pickup call

我的梦境 提交于 2019-11-29 11:54:29
I want to pickup call in Asterisk using AMI. I can originate call, but totally don't know, how to answer the phone... Script for calling: #login sock = socket.socket(af, socktype, proto) sock.connect(sockaddr) sock.send('Action: login\r\n') sock.send('Events: off\r\n') sock.send('Username: '+str(ast_server.login)+'\r\n') sock.send('Secret: '+str(ast_server.password)+'\r\n\r\n') #originate call sock.send('Action: originate\r\n') sock.send('Channel: ' + str(user.asterisk_chan_type) + '/' + str(user.internal_number)+'\r\n') sock.send('Timeout: '+str(ast_server.wait_time*1000)+'\r\n') sock.send(

How to make asterisk server automatically response to SIP call?

六眼飞鱼酱① 提交于 2019-11-28 13:09:05
问题 My objective: I want to use softphone(3CX phone) register with asterisk server, and make call to the server and asterisk act as a server to automatically response something, like play a song. How i did: I installed asteriskNow using virtualbox, and registered the softphone by setting exntension for my SIP device (extension 333). And i write a dialplan in etc/asterisk/extensions.conf . The dialplan is : [incoming] exten =>s,1,Answer() exten =>s,n,Playback(dir-intro-oper) exten =>s,n,Hangup() I