Friday, November 6, 2009

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.

Posted by solo at 07:58:20 | Permalink | No Comments »

Thursday, September 17, 2009

asterCC v0.13 beta released

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

Posted by solo at 06:20:28 | Permalink | No Comments »

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

Posted by solo at 06:19:27 | Permalink | No Comments »

Wednesday, April 15, 2009

asterCC released 0.12-beta

asterCRM new features:

  • campaign result statistics
  • agent dialed result statistics
  • support set start time and end time for campaign
  • supply a shell for backup file and database
  • agent can add a scheduler dial for a customer
  • support set dial waittime in campaign
  • agent can add a transfer link in note
  • astercrm workwith asterbilling simplely
  • supoort delete a uploaded file
  • agent pause function
  • reload or restart asterisk in web

asterBilling new features:

  • generate sip extension and sip reload by admin
  • display connect speed of clid
  • comment in credithistory
  • reloadrate, useSrcchanWhenNoClid
  • shortcut update customer rate for groupadmin
  • support payment by paypal
  • reload or restart asterisk in web
Posted by solo at 18:03:59 | Permalink | No Comments »

add callshop & realtime billing feature to your a2billing

If you have a a2billing working already, you may want to add some more features, like make it work as a hosted callshop, here we’ll introduce how to add callshop feature using asterbilling.

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

sc-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

sc-4

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

sc-1

6 login as groupadmin/operator and enjoy :)

sc-3

Posted by solo at 17:36:00 | Permalink | No Comments »

Friday, March 20, 2009

building a virtual office using astercrm ,freepbx and asterisk

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

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

freepbx_queue

and u can set some options for this reciption queue

freepbx_queue_detail

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

freepbx_trunk

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

freepbx_trunk_1

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

freepbx_inbount_route

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

astercrm_account

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

astercrm_trunk_info

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

astercrm_agent_1

when ringing

astercrm_agent_2

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.

Posted by solo at 02:07:41 | Permalink | Comments (1) »

Monday, December 15, 2008

dialer, queue and popup for asterisk callcenter(freepbx,trixbox,elastix,pbxinflash)

The latest asterCRM has a great improvement in dialer, and with asterCRM, it’s quite easy to build a call center. Here’s a how-to for a outbound call center with Freepbx and astercrm. Following by this how-to, you can creat such a solution:

asterCRM dialer will call the numbers in your diallist, and when the call is connected, it would be redirect to a queue, where your agent will answer the call and talk the customers,  they can do survey , sales or whatever you want.

* freepbx is a web gui for asterisk which is widely used in asterisk applications, like trixbox, elastix, pbx in flash …

  • install freepbx

For freepbx installation, you can read the installation document from freepbx website http://www.freepbx.org. If you are using trixbox, elastix or pbx in flash, then u can skip this, it have freepbx build in already.

  • install asterCRM, make sure asterCRM daemons (astercc and astercctools) are running

for asterCRM installation, go and check asterCRM wiki:

http://wiki.astercrm.org/index.php/AsterCRM_Installation#Using_the_install_script

  • add extensions for your agents and set a queue to receive calls from asterCRM dialer

login into freepbx, start add extensions for your agent

freepbx-add-extension

then add a queue

freepbx-queue

  • set group/user in asterCRM

next login asterCRM as admin, create group “outbound sales” and add extensions for agents you created above, go wiki for more detail

http://wiki.astercrm.org/index.php/Create_Group

http://wiki.astercrm.org/index.php/Create_Extension

make sure “Extension” matched “Outbound CID”  or “Extension”(if outbound cid is blank) in freepbx

so now all your agents should get a username/password for asterCRM.

  • set a campaign

astercrm-add-campaign

put the queue number “02″ in and check the “bind” checkbox, then as soon as customer answer the call, it would be putted into the queue, and your agent could start talk!

http://wiki.astercrm.org/index.php/Create_Campaign

  • import diallist

say you have already get 1000 numbers you want agents make call to, then use import function to import these number to diallist for this campaign

http://astercc.org/tips/2008/11/import-data-in-astercc.html

  • agents login

your agent should get ready to start! Log into astercrm and log into the queue if they are dynamic agent.

astercrm-agent

  • start dialer

both groupadmin and admin could start the dialer, go management interface and then click “Dialer” icon

astercrm-dialer

check the checkbox “start”, it will start dial, u can set limit by channel or limit by agent in the queue, in the latest version, it would not stop dial even u close the page, but will stop if u uncheck the checkbox “start”.

http://wiki.astercrm.org/index.php/Predictive_dialer

  • agents login

when a customer is connected and redirect to the agent in queue, it would popup customer information if you have in database

astercrm_agent_popup_en

Posted by solo at 09:25:32 | Permalink | No Comments »

Sunday, December 14, 2008

Survey features in asterCRM for outbound asterisk call center

In asterCRM, it provides a survey features, so you can set survey for your customers, here will give you a simple introducation for this:

first you need to add a survey, login as admin/group admin go manager interface and click survey icon, click “Add” button for a new survey, you can add several options for one survey, and each survey it provides three kinds: radio, checkbox and text

survey-1

survey-2

Click the “item” link next to the option to enter items for this option

survey-3

keep putting options and items until u finish this survey.

Then you can  put a survey from the agent page

survey-4

survey-5

*when there’s a customer or contact in your record form, if u click the “Add” link of a survey, the result will be linked with the customer

Click the “Detail” link of a survey, u can get a statistic of this survey

survey-6

Posted by solo at 09:11:02 | Permalink | No Comments »

Sunday, November 23, 2008

Rates setting in asterbilling for asterisk billing

There are three rates in asterBilling, reseller rate, callshop rate and customer rate.

  • reseller rate: the rate admin sell to reseller
  • callshop rate (group rate): the rate reseller sell to callshop (group)
  • customer rate: the rate callshop sell to customers

Rates in asterBilling could be inherited, for example, here’s two records in resellerrate

dialpreifx = 0086
number length = 0
connect charge = 0.2
init block = 60
rate = 0.2
billing block = 60
resellerid = 0

dialpreifx = 0049
number length = 0
connect charge = 0.4
init block = 60
rate = 0.4
billing block = 60
resellerid = 0

Because we dont specify which reseller is this rate for (resellerid = 0), this rate could be used for all resellers.  So what if some resellers want to change this rate rather than use this “default” rate? quite simple, just add another rate for the reseller:

dialpreifx = 0086
number length = 0
connect charge = 0.2
init block = 60
rate = 0.2
billing block = 60
resellerid = 1

so for this reseller (resellerid=1), when customer dial a number begin with 0086, it will use this new rate, we  can call it “overwrite”, but for other resellers who dont set their rate, it would use the one admin setted.

Just like reseller rate, group rate could also be inherited.

A:
dialpreifx = 0086
number length = 0
connect charge = 0.2
init block = 60
rate = 0.2
billing block = 60
resellerid = 1
grouprid = 1

B:


dialpreifx = 0086
number length = 0
connect charge = 0.4
init block = 60
rate = 0.4
billing block = 60
resellerid = 1
grouprid = 0

C:

dialpreifx = 0086
number length = 0
connect charge = 0.4
init block = 60
rate = 0.4
billing block = 60
resellerid = 0
groupid = 0

so rate C is a rate for all reseller and all group,  rate B is for all groups in reseller 1, rate A is only for group 1.

Through inherit rate, admin could be much easier to control the rates, like just set one rate for all resellers, and adjust some when they require some difference, also reseller could just set one rate for his callshops, and only do minor change to provide different rate plan.

Posted by solo at 04:57:55 | Permalink | No Comments »

asterbilling and asterisk2billing (a2billing) for asterisk billing

a2billing is a widly used billing system, so what’s the difference between a2billing and asterbilling?

system theory:

a2billing work with asterisk through AGI, each call would be handled by a2billing.agi, so you need change your astierks context so that when your phone make calls it will go to a2billing.agi, of course you need to copy a2billing scripts to your asterisk server.

asterbilling runs as linux daemons, connect to asterisk via AMI over tcp,  no need change anything on your asterisk, asterbilling can bill it. Even asterbilling could work with a2billing.

license & free:

a2billing: 100% open source and free to use.

asterbilling: only web scripts are open source, and provides 5 free simultaneous channels, have to purchase when need more channels.

performance:

a2billing: like 100 simultaneous calls on single server? i didnt test, :(

asterbilling: passed 240 simultaneous calls testing, but didnt test more.

best usage:

a2billing: calling card, callback or wholesale solutions

asterbilling: billing for embedded astiersk, pbx (like all freepbx based system), callshop, hosted callshop soltution

other keywords:

a2billing: openser

asterbilling: realtime billing, reseller

Posted by solo at 04:55:33 | Permalink | No Comments »