After elected “login”, then the current agent has been logged into the queue which belongs to the queue number, the queue was according to the state of all the agent be logged to assign the calls. After elected “pause”, then the agent has been logged are not be assigned phone calls whether he is busy or free. When elected logoff, the current agent were logged off  from the queue, the queue will not assign any phone calls to the agent.

Return to the management interface, enter the dial list below campaign find the queue number corresponding to the campaign name, click on the campaign “edit” to open.

Find queue number, indicate the agent to be logged in the queue number; Queue Context, the crew used the queue context, freePBX use the latest version is the current from-queue; Use Extension Channel For Queue, where there are two cases. One case is logged in default, do not be checked before “Use Extension Channel For Queue”, in the asterisk CLI to perform queue show.

Another case is to use” Use Extension Channel For Queue”, checked before “Use Extension Channel For Queue”, in the asterisk CLI to perform queue show.

Back to the management interface, enter the extension to find the user name of current agent, click “edit”

Channel option, sip/8003, 8003 is the current agent extension.

1. config a Queue in freepbx

usually you want to config a queue to resonpse customer if they want to reach some live agent, so we config a queue first

2. config an IVR in freepbx

2.1 first we will add some voice in the IVR, you’d like to use a recording software, just notice that in asterisk, it requires to use wav format and 16bit, 8000HZ mono

2.2 add a recording

you can either upload the recording you finished in your pc, or use a ext. in system to record a new one

2.3 add IVR

now queue and recording is ready, we could add the IVR

in announcement, we select the recording we just done, and also we added two options to accept customer input, when customer hits 1, he will go queue 900, and when hits 2, he will go to ext. 5001.

4. add misc

then we need add a misc so the dialer could reach the IVR

remember this feature code 800, we will use it when configing campaign in astercc, you could use some other code as you like, just make sure it’s unique in your freepbx

now we finished the job in freepbx, you’d like to dial 800 from any extension, it supposes to bring you to the IVR.

5. config the campaign in astercc

login as admin, and go to campaign, add a campaign as following

make sure in “Inexten” you put the same code as in “misc application”, and in “Queue number” put the same queue number as your freepbx queue

6. add agent user in astercrm

you’d like to add some account for your agents if you want them get a popup form when they start answer customer calls, go to “Extension” to add astercrm account for your agents, make sure the account “extension” match the ext. in your freepbx

7. start dialer and test agent

before start dialer, you’d like to check your astercc.conf

make sure this parameter is configed as

doublecheckcampaign = yes

then restart astercc daemons

/opt/asterisk/scripts/astercc/asterccd restart

ask your agent login and as admin go to dialer page, start the dialer

as an agent, he will get a popup when customer answered the call and hit 1 for queue

8. check report

you could go to dialedlist to check the campaign status

* to get a working freepbx and asterisk easily, you could choose  astercc box, elastix, pbxinaflash or trixbox

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.

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.

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

then add a 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

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.

• start dialer

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

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

benefits of asterBilling hosted callshop solution:

• reseller, callshop, customer three level billing
• all web based
• high performance
• all asterisk system compatible

here, i’ll introduce u how to build a hosted callshop solution using asterisk and asterbilling.

1. step1, install asterisk

2. step2, set trunk and dialplan in asterisk

edit /etc/asterisk/sip.conf and add your trunk there

then set dialplan, go to /etc/asterisk/extensions.conf and add a context there

3. step3, install asterBilling

4. step4, check asterbilling.conf.php

We need to modify asterbilling config file to meet our system, so check the “asterbilling.conf.php” in asterbilling folder, find section “sipbuddy” change context to be “context = from-booth”, so the sip peer generated by asterbilling will use context “from-booth” for outbound calls.

* if you are using freepbx, you can use “context=from-internal” here, then it you can set outbound in your freepbx and all booth will use that.

By default, asterbilling will generate all sip peers to the file “/etc/asterisk/sip_astercc”, you can change to other name if you want, or leave it blank if u dont want asterbilling generate the sip peers

also we need to include the conf file in sip.conf so that asterisk could load peers asterbilling generated

modify /etc/asterisk/sip.conf and add

#include sip_astercc.conf

5. step5, set resellers and groups

6. step6, add clid as reseller

7. step7, set rates

asterbilling provides three level billing:

rate to reseller: the rate you sell to resellers

rate to callshop: the rate resellers sell to callshops

rate to customer: the rate callshops sell to customers

8. step8, login as groupadmin/operate check callshop interface

9. step9, check reports

• Trixbox
• Elastix
• Callweaver
• Freepbx
• Magiclink
• Fonesoft
• asterisk2billing (a2b)
• pbx in a flash
• …

so when you are using a asterisk based system and want to add call center or billing features, asterCC is a good choice.

asterCC solutions connect to your asterisk via AMI over tcp, so even a embedded asterisk equipment would use asterCC for billing or contact center.

asterCC works based callerid, so it doesnt care what’s the asterisk dialplan or how a agi work, as long as you have  correct callerid in your asterisk, asterCC could work with it.

Working as daemon service in linux, asterCC is stable and extremely efficient, we have test that it could support at least 240 simultanieous asterisk calls.

Open source as all web scripts is, there’s also the possibility that you make your own solution based astercc daemons, and with the 5 free simultanieous channel license it provides by default, no need to pay a dollar for small business, like to bill an asterisk pbx with users less than 12.

the first thing is, you must have freepbx installed and have a user their, say you want to bill these two users: solo <8000> and donnie <8001>

next go to asterBilling manager login (like http://asterccserver/asterbilling/manager_login.php) and login as “admin”

Although we only want to bill our pbx,  still have to add a reseller first, go to “reseller” and click “Add” button, pur some message in the form and click “continue”

i left “Credit Lmit” to be null and “Limit Type” to “No limt” coz i just want to know how much each phone dialed each month.

then go to “Account Group” and add a group there

next go to “Clid” and add your phones, write down the “Pin” of the phone.

ok, now you have added the phones you want to bill in asterBilling, we need to add rate next, we have reseller rate, group rate and customer rate, I want to know the phone cost of each phone, so i only set customer rate

go to “Rate to Customer” and add a rate

this 0049 rate mean all calls begin with 0049 will be charged as such rate, 0.2 for the first 180 seconds and 0.2 each minutes after the first 180 seconds, it’d be charged every 60 seconds.

We have a rate which prefix is “default”, so this means if no prefix matched, the call will use this rate, we can use this for internal calls.

Also you may noticed that we didnt select Reseller and Group, this means all Reseller and Group will use this rate.

After all rates are added, your configuration job is done, before you leave asterBilling, make sure astercc daemons is working or else just start them by “asterccd start”

When a phone user want to check his CDR and how much he have dialed, he can login asterBilling via phone login page, like http://asterccserver/asterbilling/login.php, login use phone number and pin

then you can see your CDR history and how much each call cost:

First, we need to add some extensions and queues in FreePBX:

add extensions:

go to FreePBX extensions page,

then we add a queue and choose a dynamic agent for it like following figure

yes, if you want to add a dynamic agent for a queue, you should add a agent number with A in queue agent list, and we can notice that message from FreePBX.

so we need to counfigure agent.conf in asterisk conf directory, here we add a dynamic agent which number is 1000, its  passoword is 0000, name is Brad

then we add a dialplan for agent login in extensions_custom.conf like following line:

update the context of extension 8000 to “agent-test”.

here, we had a queue which include a dynamic agnet, well then we need add a user in astercrm, go to the extension management page of astercrmand add a new extension and assign agent 1000 to him:

Agnet start work:

first the extension have to login as a agent to PBX, when 8000 dial “*789″, it would hear the prompt that require to enter the agent number and password to login as the agent, enter agent number 1000 and password 0000, it can login as agent 1000.

login to asaterCRM by brad/1234,  the asterCRM could pop-up if anyone call to Agent/1000 or extension 8000.