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

asterCC V0.14-beta-rc1 (378) - 14.84 MB

asterCC-BOX-0.14-beta-rc1

asterCRM 0.62:

* added dnc(do not call list) for daillist

dnc1

dnc2

* 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

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asterCC and asterCC-BOX released 0.14-beta

asterCC V0.14-beta (327) - 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)

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

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

crd_amd_en

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

dialist_en

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

dialist_en1

dialist_en1

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

dialist_en2
dialist_en3

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config trunk and failover trunk in asterbilling

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

extensions_astercc.conf

b)config the trunk for reseller

reseller_trunk1

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

reseller_trunk2en

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.

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new astercc released

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.

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asterCC & asterCC BOX released 0.13

asterCC V0.13 (3048) - 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_agentsettings

astercrm agent management
astercrm_clearscreen

astercrm clearscreen
astercrm_dialliatpannel

astercrm diallist pannel
astercrm_panelswitcher

astercrm panels witcher
astercrm_mp3player

astercrm mp3player for recording files
asterbilling_professional

asterbilling professional mode
asterbilling_portuguese

asterbilling portuguese language support
.
freepbx2.6 in asterCC BOX 0.13

freepbx2.6 in asterCC BOX 0.13
asternic_realtime

asternic_realtime
asternic_distribution

asternic_distribution


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tutorial: use astercrm & asterisk for broadcasting

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

context

3. add group in astercrm

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

group

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

campaign

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

numbers

import:

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

dialer

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

worktime

then create a work time package and add the worktime in

worktime_package

then set the campaign to use this work time package

campaign_with_worktime

8. check dial result

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

dialedlist

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.

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auto recycle the dialed number in a campaign

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.

campaign

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asterCC v0.13 beta released

asterCC v0.13beta (839) - 11.53 MB

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

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

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

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