Freepbx Archives - Call Center Software Solution Provider | AsterCC

asterCRM queue login and pause function make dynamic agent better serve the inbound and outbound in call center

By | asterCRM | No Comments

Log in the system page, enter the agent interface. Find the campaign pannel(Queue ) below the campaign corresponding to the queue number, and the two options, login, pause.

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.

tutorial: use astercc,freepbx and asterisk to build a broadcasting system with IVR and Queue agents

By | asterCRM | No Comments

the most feature in astercc is the predictive dialer, using the dialer you could improve the work efficiency. In this tutorial, I will introduce how to setup a dialer with a pre-configed IVR:  when dialer start work, customers will hear a IVR which you configed in Freepbx, so we could also config in the IVR to accept customer input, and we can forward to an new IVR or agents in a queue.

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

config trunk and failover trunk in asterbilling

By | Tutorials | No Comments

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.


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)



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.

building a virtual office using astercrm ,freepbx and asterisk

By | asterCRM | 6 Comments

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


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.

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

By | asterCRM | 16 Comments

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:


  • 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



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.

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asterbilling hosted callshop solution for asterisk

By | asterBilling | 22 Comments

asterBilling is a realtime billing software for asterisk. Through asterBilling, it’s very easy to build a hosted callshop solution for asterisk. 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 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 callshop 9. step9, check reports

why we say asterCC solutions could work with all kinds of asterisk based solution.

By | asterBilling, asterCRM, asterisk | No Comments

asterCC package provides a call center solution and a billing solution for asterisk, the most important feature is, asterCC could work with all asterisk based solutions and no need do any modification to the original system,  as we have tested, including:

  • 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.

How to set asterBilling working with Freepbx, Trixbox, Elastix …

By | asterBilling | 21 Comments

asterBilling is a realtime billing solution for asterisk, could be used as a hosted callshop or just a simple billing system for your asterisk pbx. Many people are using freepbx based system as their pbx, like trixbox, elastix … so here i’ll introduce you how to use asterBilling to bill your asterisk pbx.

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

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Dynamic Agent mode with asterisk, FreePBX and asterCRM

By | asterCRM | No Comments

Dynamic Agent mode is very useful in a outbound call center, agents login to a queue and hear some music we definied in the queue, then predictive dailer start works, it dials customer numbers and once it get connected the dialer will redirect the call to the queue, and agent in the queue could hear a beep immediately then could start talking to the customer, it’s so quick that customer just think it’s a call from the agent. With asterCRM, agent could get customer information once the call get connected, here we’ll introduce how to set dynamic agent in freepbx and asterisk: 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.