asterCC, asterisk callcenter and billing solution

set the special dialplan work with campaign callerid

admin — Wed, 25 Aug 2010 08:10:18 +0000

when you set a callerid for a campaign in astercrm, maybe you could find that the campaign callerid is not only show in customer’s phone , but also in agent’s phone. Now let’s fix this poplem together with asterisk dialplan.

First,we set a new dialplan such as following lines:

[from-internal-mask]

exten => _X.,1,NoOp(${CUSCID})
exten => _X.,n,GoToIf($["${CUSCID}" = "NONE"]?dial)
exten => _X.,n,Set(CALLERID(num)=${CUSCID})
exten => _X.,n(dial),GoTo(from-internal,${EXTEN},1)

In above dialplan we said you are using from-internal for dialout, if not you should change “from-internal” to your dialplan at last line.

And then, we should edit the ‘Incontext’ of campaign to ‘from-internal-mask’

Now, try it if work well.

asterCC-1.0-beta-rc1 released (commercial version)

admin — Mon, 16 Aug 2010 09:50:22 +0000

astercc1.0 is a contact center system, while providing a powerful CRM (customer relationship management) functions, including features we often use like inbound / outbound missile screen, call recording, interactive voice menu (IVR), call queues and so on. Basicly by using astercc, you can record every customer call, the time when the customer calls appear on the screen of the customer contact information and historical contact records, or the design of an interactive voice menu (IVR) to provide self-service inquiries. In addition, the use of reporting tools, so you can quickly call from thousands of the critical information. Finally, astercc also offers a variety of practical tools, such as the task of setting and reminders by telephone, SMS and EMAIL of enterprise information, check the map or use the google map paths.

If you already have your own business system, you can come in its perfect integration¹⁾, for example, a customer calls you want your system to show customers information, used astercc the secondary development interface, you can quickly complete the work.

For ISP, astercc is the best choice to carry out new business, whether functional or telephone call center switching (PBX), astercc support SAAS mode, simply building a system, we can provide services to multiple customers .

asterCC 0.14-beta and asterCC BOX 0.14-beta-rc1 released

admin — Thu, 17 Jun 2010 09:05:22 +0000

asterCC V0.14-beta-rc1 (631) - 14.84 MB

asterCC-BOX-0.14-beta-rc1

asterCRM 0.62:

* added dnc(do not call list) for daillist

* fixed worktimepackage bug with astercctools
* fixed can not update callresult to dialedlist in astercrm.agi
* added update groupid and accountid to mycdr(parameter “update_groupid” in astercc.conf)

asterBilling 0.12:

* updated astercc daemon to fixed the a billing issue, which may cause the billing sec is less than the real billing time

* added playbalance.agi for listen the balance of callshop by a clid

asterCC and asterCC-BOX released 0.14-beta

admin — Thu, 27 May 2010 02:40:14 +0000

asterCC V0.14-beta (398) - 14.79 MB

asterCC-BOX-0.14-beta

asterCRM 0.062:

* added export in cdr page

* added parameters(uniqueid/calldate) when ust extenal crm

* not display diallist function when login as dynamic agent

* added update groupid and accountid to mycdr(parameter “update_groupid” in astercc.conf)

* improved processmonitors

* improved export funciton, supoorts to export xls and cvs format

* improved report page

* astercc daemon support astersik 1.6.x

* added script astercrm_update_cdr(update customer id to mycdr)

* added astercrm.agi(update call result to dialedlist and Answering Machine Detect when using predictive dialer)

* fixed worktime bug when use predictive dialer

asterBilling 0.12:

* improved profile page of resller and groupadmin(could view the own balance)

* fixed the currency bug of recharge by paypal

* added display callshop Balance in systemstatus

* improve clid page (could control the clid if display in systemstatus as a booth)

* improved systemstatus(could be open manager page in current page)

* added set free call function in receipt page

* improved receipt to close page when paid

* improved cdr page to set special color for unbilled or free call

* improved report(display memo and note)

* added could set a special turnk for each resller dialout

* astercc daemon support astersik 1.6.x

* improved reselleroutbound.agi, support spare trunk

* added account_log page(records the account login logs)

Use astercrm.agi to answering machine detect and auto update call result of diledlist

admin — Thu, 27 May 2010 02:26:18 +0000

In astercrm 0.062，we call use astercrm.agi to answering machine detect and auto update call result of diledlist.

How to:

a) in your asterisk, add two context in your dialplan for reselleroutbound.agi, in directory scripts, we also provided a conf file named “extensions_astercc.conf”, [from-astercrm] and [from-astercrm-amd] is the context for astercrm.agi, if you have installed astercc via the shell script install.sh, this conf file will be copy to your asterisk etc folder, and add a new line in your extensions.conf “#include extensions_astercc.conf”, now you can use [from-astercrm] and [from-astercrm-amd] as the context for astercrm.agi, 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) configure campaign

select “Call Result Detect” and “Aswering Machine Detect”, and then fill the context for them

c) try it we add some records in diallist for campaign

dialout use predictive dialer: this one , the callee refulsed, got the following data in dialedlist:

dialist_en1

and this one the callee is a fax machine, got ollowing data in dialedlist and surveyresult:

config trunk and failover trunk in asterbilling

admin — Thu, 27 May 2010 02:25:55 +0000

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.

new astercc released

admin — Sun, 25 Apr 2010 09:51:29 +0000

the new generation of astercc release its first version 1.0 beta, can be download on http://astercc.org/downloads, the biggest difference of this new product is that it’s not open source any more, but more useful & powerful features, for details please visit http://wiki.astercc.com. Also, we will keep developing astercc 0.x, in fact we will release 0.14 soon.

asterCC & asterCC BOX released 0.13

admin — Thu, 19 Nov 2009 12:26:51 +0000

asterCC V0.13 (3193) - 14.26 MB

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

tutorial: use astercrm & asterisk for broadcasting

admin — Fri, 06 Nov 2009 06:39:23 +0000

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

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.

auto recycle the dialed number in a campaign

admin — Fri, 30 Oct 2009 03:57:17 +0000

in a outbound campaign, some number would be failed to be reached, like no answer, hangup caused by bad voice quality, so we need to dial these numbers again, then we provide the auto recycle feature.

Max trytime: the maximum time we will try, if we have dialed the number more than a number, it would not dial any more

Recyle time: when to recyle the number, 3600 means it will recycle the number after 3600 seconds if it dialed last time.

Min Duration: if the talking time is equal or less than the min duration, it will be recycled.