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 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

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