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
1.upgrade database
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 new features:
asterBilling new features:
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
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 
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.
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 …
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.
for asterCRM installation, go and check asterCRM wiki:
http://wiki.astercrm.org/index.php/AsterCRM_Installation#Using_the_install_script
login into freepbx, start add extensions for your agent
then add a queue
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.
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
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
your agent should get ready to start! Log into astercrm and log into the queue if they are dynamic agent.
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
when a customer is connected and redirect to the agent in queue, it would popup customer information if you have in database
a2billing is a widly used billing system, so what’s the difference between a2billing and asterbilling?
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
astercc works as a daemon in linux, it get events from asterisk AMI and store the CDR to two mysql table.
the first table is “curcdr” which will store all simultaneous calls, when call starts there would be a new record in the table, and when it stops, the record would be removed to another table “mycdr”
the record could be:
mysql> select * from curcdr;
+——+——+———-+—————+—————–+———————+———————+—————–+—————–+————-+———+——–+——–+—————-+—————-+————-+
| id | src | dst | srcchan | dstchan | starttime | answertime | srcuid | dstuid | disposition | groupid | userid | credit | callshopcredit | resellercredit | creditlimit |
+——+——+———-+—————+—————–+———————+———————+—————–+—————–+————-+———+——–+——–+—————-+—————-+————-+
| 3880 | 8807 | 8806 | SIP/8807-6fc2 | SIP/8806-d962 | 2008-02-28 12:26:01 | 2008-02-28 12:26:10 | 1204172801.3813 | 1204172801.3814 | link | 0 | 0 | 0.0000 | 0.0000 | 0.0000 | 0.0000 |
| 3881 | 8000 | 84350822 | SIP/8000-1684 | SIP/trunk1-a73d | 2008-02-28 12:26:13 | 0000-00-00 00:00:00 | 1204172812.3815 | 1204172813.3816 | dial | 0 | 0 | 0.0000 | 0.0000 | 0.0000 | 0.0000 |
+——+——+———-+—————+—————–+———————+———————+—————–+—————–+————-+———+——–+——–+—————-+—————-+————-+
and the record in mycdr could be
mysql> select * from mycdr order by id desc limit 0,2;
+——+———————+——+———-+—————+—————–+———-+———+————-+————-+———–+—————–+—————–+———-+——–+—————-+—————-+———+——–+
| id | calldate | src | dst | channel | dstchannel | duration | billsec | disposition | accountcode | userfield | srcuid | dstuid | calltype | credit | callshopcredit | resellercredit | groupid | userid |
+——+———————+——+———-+—————+—————–+———-+———+————-+————-+———–+—————–+—————–+———-+——–+—————-+—————-+———+——–+
| 3682 | 2008-02-28 12:26:13 | 8000 | 84350822 | SIP/8000-1684 | SIP/trunk1-a73d | 5 | 0 | NO ANSWER | | UNBILLED | 1204172812.3815 | 1204172813.3816 | | 0.0000 | 0.0000 | 0.0000 | 0 | 0 |
| 3681 | 2008-02-28 12:24:40 | 8806 | 8807 | SIP/8806-c36f | SIP/8807-0139 | 8 | 6 | ANSWERED | | UNBILLED | 1204172719.3811 | 1204172720.3812 | | 0.0000 | 0.0000 | 0.0000 | 0 | 0 |
+——+———————+——+———-+—————+—————–+———-+———+————-+————-+———–+—————–+—————–+———-+——–+—————-+—————-+———+——–+
2 rows in set (0.00 sec)
so if you want to get real time cdr datas from your asterisk for your application, you can try astercc, it provide 5 free channels, means it could catch 5 simultaneous calls in your asterisk, it has been tested that astercc could catch as much as 240 simultaneous calls.
astercc daemon could be download from http://sourceforge.net/project/showfile … _id=202441, if you only want to get the calls information, you just need astercc and astercc.conf in directory daemon in the package
