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

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freepbx2.6 in asterCC BOX 0.13

asternic_realtime

asternic_distribution

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.

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.

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:

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

unzip the package u could see folder “sql”

files in asterCC 0.12 should be:

astercc0.1b-0.1.sql
astercc0.1-0.11.sql
astercc0.11-0.12b.sql
astercc0.12b.-0.12sql
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 one by one, then u get database of v0.12

2. stop astercc daemons

/opt/asterisk/scripts/astercc/asterccd stop

3. cp the new html & daemon files

4.  set conf files

files include astercc.conf, asterbilling.conf.php, astercrm.conf.php

5. start astercc daemons

/opt/asterisk/scripts/astercc/asterccd start

6. login and check

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

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

2.  add a queue for your receiptions which would be used to answer incoming calls, we only add 8000 and 8001 in this queue

and u can set some options for this reciption queue

3. add a trunk which could be used for incoming calls

and the most important, set registry for this trunk so that u can get calls in

4. add a inbound route so that the receiption queue could answer your incoming calls

now make a call to your DID number, if everything is allright, phones of receiption should ring

5. go to astercrm and add account for your receiptions

6. add trunkinfo so your receiption could get some information about the number customer dialed

here Trunk Channel should be the username of your trunk, not trunk name in freepbx

7. login as a receiption accound and try make a call

when ringing

when talking

this tutorial could be used on trixbox, elastix or any other system using freepbx, also u can config receiption account and dialplan by your self.

say admin set three rates for reseller:
default     0.2        for all resellers
0086        0.25     for reseller1
00852     0.3        for all resellers

then reseller1 logged in and want to add sell rate for his callshop

1. we add rate for North Americe, which prefix is 001, it will give the buy rate for 001 after input box (we didnt set prefix 001 in rate to reseller, so it will use default rate)

2. we add rate for China, which prefix is 0086, it will give the buy rate for 0086 after input box

3. we add rate for HK, which prefix is 00852, it will give the buy rate for 00852after input box

also, when groupadmin add customer sell rate, he can also see his buy rate from reseller

1. add a new conf in your a2billing

add a new conf like [agi-conf2] in a2billing.conf, make sure you have the changed the following options:

; Manage the answer on the call

answer_call = NO
play_audio = NO
use_dnid = YES
number_try = 1
say_balance_after_auth = NO
say_balance_after_call = NO
say_rateinitial = NO
say_timetocall = NO
cid_enable = NO
cid_auto_assign_card_to_cid = NO

anyway, disable all prompt & announcement

2. add new dialplan in asterisk extensions

by default, sip peer generated by a2billing will use context a2billing, so we add

[a2billing]
; for asterbilling booth
exten => _X.,1,DeadAGI,a2billing.php|2

3. add costomer in a2billing

then we add a customer in a2billing, make sure you enabled sip or iax account, then click the “generate” button and click “reload” link

also u may want to set this customer as “postpay” and a big number for the limit coz you would not charge customer in a2billing, just make sure this customer could make calls with no problem

4. set your ip phone

go to “List Sip-friend” or “List iax-friend” get the username/secret for your phone, then try make a call, if everything goes well, u should make a call successfully

5. add clid in asterbilling

go to asterbilling and create clid using the username(if there’s callerid defined for this customers, use callerid instead) in sip-friends

6 login as groupadmin/operator and enjoy