Page 16 – Call Center Software Solution Provider

May 27, 2010

1

config trunk and failover trunk in asterbilling

By solo@astercc.org | Tutorials | No Comments

in the new asterbilling, we provide a reselleroutbound.agi, so you can specific different reseller use different trunk (ex. each reseller use a account in a2billing), and you can config a failover trunk for the reseller.

Howto:：

a) in your asterisk, add a context in your dialplan for reselleroutbound.agi, in directory scripts, we also provided a conf file named “extensions_astercc.conf”, [asterbilling- outbound] is the context for reselleroutbound.agi, if you have installed astercc via the shell script install.sh, this conf file will be moved to your asterisk etc folder, and add a new line in your extensions.conf “#include extensions_astercc.conf”, now you can use [asterbilling-outbound] as the context for asterbilling, if you are using asterCC-Box, it’s configed already. If you installed astercc manually, you would like to copy extensions_astercc.conf to asterisk etc folder(usually it’s /etc/asterisk ), and add the include line in your extensions.conf (#include extensions_astercc.conf, if it’s a system based freepbx, please add this line to /etc/asterisk/extensions_custom.conf)

extensions_astercc.conf

b)config the trunk for reseller

when clid dialout, it’ll use turnk1 first and if dail failed, it’ll try to dial by trunk2

There are three type of trunk: auto，default and customize

auto：reselleroutbound.agi don’t proccess anything，and goto next step of context

default：your can select a default trunk that set in [resellertrunk] segment asterbilling.conf.php, could be set tow default system trunk:

[resellertrunk] trunk1_type = sip
trunk1= reselleroutbound1
trunk2_type = sip
trunk2= reselleroutbound2

customize：add new trunk for this reseller，should click “reload” button to generate asterisk conf file when saved trunk infomation

when you add the trunk for the first time, when you reload, if will have two conf file: sip_astercc_registrations.conf and sip_astercc_trunks.conf , if you are not using astercc-box, please include these files to your sip.conf(for freepbx based system, please add #include sip_astercc_registrations.conf to /etc/asterisk/sip_registrations_custom.conf, and add #include sip_astercc_trunks.conf to /etc/asterisk/sip_custom.conf, and then do sip reload in asterisk , for the next time you add trunk, just need click the “reload” button.

April 25, 2010

1

new astercc released

By solo@astercc.org | Latest News | No Comments

astercc 1.0 released

asterCC & asterCC BOX released 0.13

By solo@astercc.org | Latest News | No Comments

[download#14#size]

asterCC-BOX-0.13 download

asterCC BOX 0.13:

* updated to freepbx 2.6 rc2
* updated to asternic 1.2
* updated to asterCC 0.13

asterCRM 0.061:

* added agents manager in astercrm to manage agents.conf
* fixed the bug that cant edit worktime_package
* added callOrder field in diallist
* added diallist panel in portal page
* added the daemon to convert recording files to mp3 format
* added mp3 online player
* added agent portal panel switcher
* added clear screen button in agent portal

asterBilling 0.11:

* fixed the prefix billing
* added professional mode
* added member mode switch
* added Portuguese support

astercrm agent management

astercrm clearscreen

astercrm diallist pannel

astercrm panels witcher

astercrm mp3player for recording files

asterbilling professional mode

asterbilling portuguese language support

.

freepbx2.6 in asterCC BOX 0.13

asternic_realtime

asternic_distribution

November 6, 2009

1

tutorial: use astercrm & asterisk for broadcasting

By solo@astercc.org | asterCRM | 6 Comments

in this tutorial, it will guide u how to broadcast your message in asterisk and astercrm.

1. add outbound context in asterisk

add the following content in your dialplan (like extensions.conf)

[for-outbound] exten => _X.,1,Dial(SIP/yourtrunk/${EXTEN},45)
exten => _X.,n,Hangup

exten => h,1,NoOp(${DIALSTATUS})
exten => h,n,Hangup

here “yourtrunk” should be defined in your sip conf file, or you can use other trunk you have, like IAX2, ZAP, DAHD I…

2. add inbound context in asterisk

add the following content in your dialplan (like extensions.conf)

[for-collection] exten => _X.,1,NoOp(${EXTEN})
exten => _X.,Background(YOURMESSAGE)
exten => _X.,n,Hangup

exten => 1,1,Queue(1000); means when customer press 1 when it’s playing, he will reach your queue 1000

exten => h,1,Hangup()

then it will look like

3. add group in astercrm

login astercrm as admin, then go to extension->group admin, add a group for this broadcasting project

4. add campaign in astercrm

then go to diallist->campaign, add a campaign, in outcontext and incontext, we will put the context we added before, for-outbound and for-collection

5. upload the diallist

you can upload a excel/cvs file to diallist, or you can insert record to diallist table using your script

numbers.csv

import:

6. start the dialer

then u can go to dialer page to enable the campaign, also you can set a limitation of the max outbound calls there

7. set a time limitation

if you only want it dial at spcific time, you can add a time package for the campaign. first add some time

diallist -> worktime

then create a work time package and add the worktime in

then set the campaign to use this work time package

8. check dial result

go to diallist -> dialedlist, you can find the result

hope this post can help you create ur first broadcasting campaign, and u can also improve on this, like u can use a script to insert to diallist automaticly or set some survey so customer can press in their option when listening to your message.

October 30, 2009

0

auto recycle the dialed number in a campaign

By solo@astercc.org | asterCRM | No Comments

how to auto recycle in a outbound campaign

asterCC v0.13 beta released

By solo@astercc.org | Latest News | No Comments

[download#13#size]

asterCRM 0.06:

* improved survey export feature
* add a switch to control if need close all popup window after a survey
* improved dialer
* added table campaignresult
* added survye <-> campaign connection
* popup survey directly when only one survey enabled
* added surveyresult.agi, can be used to update survey when use AMD
* added new parameters which is used to control cdr data (in table mycdr)
* allow add customer name or add customer connection when import diallist, also added diallist popup
* monitor features was moved to daemon astercc
* add queuestatus page, to display realtime queue status
* fixed the bug that sort only work in the first page

asterBilling 0.1:

* fixed the billing bug when num length and prefix confilict

queue status:

queue status

July 15, 2009

4

why i cant see booth history when can see the calling call?

By solo@astercc.org | asterBilling | 2 Comments

some customers find that they can see live booth calls(screen 1) but when call is done, nothing appears in the booth box(screen 2).

live call in booth window

nothing in booth box

this happens coz the admin set sip account in “clid” when it should be “caller id” check table “mycdr” you will find that the “src” filed would be a number which doesnt match with “channel” field to fix this, just go to “clid” in asterbilling and change the clid to be the number in src field

June 27, 2009

4

how to upgrade astercc

By solo@astercc.org | asterBilling, asterCRM | 2 Comments

upgrade database unzip the package u could see folder “sql” where we put all database files, in asterCC 0.X it would looks like: astercc0.1b-0.1.sql astercc0.1-0.11.sql astercc0.11-0.12b.sql astercc0.12b.-0.12sql astercc0.13-0.14b.sql astercc….0.X.sql astercc.sql say you are using 0.1b now, so you have to execute astercc0.1b-0.1.sql, astercc0.1-0.11.sql, astercc0.11-0.12b.sql, astercc0.12b.-0.12sql until astercc…0.X.sql, then u get database of v0.X the command line for this is like mysql -uroot -p astercc < astercc….0.X.sql
stop astercc daemons /opt/asterisk/scripts/astercc/asterccd stop
cp the new html & daemon files, please notice that you may want to backup astercrm.conf.php, astercc.conf and asterbilling.conf.php first example: cp astercrm /var/www/html -rf cp asterbilling /var/www/html -rf cp scripts/* /opt/asterisk/scripts/astercc * from astercc-0.21, the package includes both 32bit and 64bit scrips, you will copy the scripts out of 32/ or 64/ to your daemon folder (usually /opt/asterisk/scripts/astercc)
config conf files check your astercc.conf, asterbilling.conf.php, astercrm.conf.php, make sure you have the correct config
start astercc daemons /opt/asterisk/scripts/astercc/asterccd start
check crontab we provides some cron script to deal with recording files and CDRs, so open README, make sure you have the scripts configed in your crontab
login web and check if any errors

June 25, 2009

3

asterCC v0.12 released

By solo@astercc.org | Latest News | 6 Comments

asterCRM 0.059:

impoved send request by javascript in portal interface
fixed can not order in customer,diallist and dialedlist page
fixed can not export in note ,diallist, dialedlist, campaign,contact
fixed can’t find astercc license file when is not running in ‘/opt’
fixed the start check of predictive doesn’t work in IE7
fixed can not record wher predictive

asterBilling 0.099:

fixed can’t display report of today
fixed bug in flash report
add check credit reseller and callshop realtime when booth calling
impoved send request by javascript in callshop interface
fixed don’t update blance when cannel limit in booth
add delete files what have uploaded
fixed can’t find astercc license file when don’t run in ‘/opt’
fixed ASR and ACD both are ‘0’ in report grid

March 19, 2009

2

building a virtual office using astercrm ,freepbx and asterisk

By solo@astercc.org | asterCRM | 6 Comments

In a virtual office, you will have few receiption but they can answer calls for hundred company, in such case, they should know which number customer dialed so that they dont mess up the calls, now we introduce u how to build a virtual call center using astercrm & asterisk.

1. add extension for receiption

open your browser and go to freepbx, click extension on left menu and add extensions for your receiption, here we have three extensions: 8000, 8001 and 8888

freepbx_extensions