[download#13#size]
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:
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).
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
- upgrade database unzip the package u could see folder “sql” where we put all database files, in asterCC 0.X it would looks like: astercc0.1b-0.1.sql astercc0.1-0.11.sql astercc0.11-0.12b.sql astercc0.12b.-0.12sql astercc0.13-0.14b.sql astercc….0.X.sql 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 until astercc…0.X.sql, then u get database of v0.X the command line for this is like mysql -uroot -p astercc < astercc….0.X.sql
- stop astercc daemons /opt/asterisk/scripts/astercc/asterccd stop
- cp the new html & daemon files, please notice that you may want to backup astercrm.conf.php, astercc.conf and asterbilling.conf.php first example: cp astercrm /var/www/html -rf cp asterbilling /var/www/html -rf cp scripts/* /opt/asterisk/scripts/astercc * from astercc-0.21, the package includes both 32bit and 64bit scrips, you will copy the scripts out of 32/ or 64/ to your daemon folder (usually /opt/asterisk/scripts/astercc)
- config conf files check your astercc.conf, asterbilling.conf.php, astercrm.conf.php, make sure you have the correct config
- start astercc daemons /opt/asterisk/scripts/astercc/asterccd start
- check crontab we provides some cron script to deal with recording files and CDRs, so open README, make sure you have the scripts configed in your crontab
- login web and check if any errors
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
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.
we provide a new feature in asterbilling, you can read your buy rate when add a sell rate manually.
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
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.
- 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
- 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
- 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
- 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
- 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
- login as groupadmin/operator and enjoy 🙂
some customers find that they get a problem when import, it just gives a XML error, that’s because there’s too much data and no enough memory for php. find the php.ini for your apache line: memory_limit it could be like 16M or 20M, try change it bigger, say 64M so it looks like memory_limit = 64M save and exit, remember restart your apache before give the try.
asterCC-BOX includes CentOS linux, it’s an easy to install contact center and billing system, based on asterisk PBX and asterCC software package, inegrated FreePBX(asterisk web GUI) with simplified chinese language package and asternic-stats(analyze your Asterisk PBX queue_log). asterCC-BOX provides flexible and stable phone system, contact center and billing(callshop solution) system for your business.
