Tag

callshop

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.

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

extensions_astercc.conf

b)config the trunk for reseller

reseller_trunk1

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

reseller_trunk2en

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.

asterCC & asterCC BOX released 0.13

By | Latest News | No Comments
[download#14#size]

asterCC-BOX-0.13 download

asterCC BOX 0.13:

* updated to freepbx 2.6 rc2
* updated to asternic 1.2
* updated to asterCC 0.13

asterCRM 0.061:

* added agents manager in astercrm to manage agents.conf
* fixed the bug that cant edit worktime_package
* added callOrder field in diallist
* added diallist panel in portal page
* added the daemon to convert recording files to mp3 format
* added mp3 online player
* added agent portal panel switcher
* added clear screen button in agent portal

asterBilling 0.11:

* fixed the prefix billing
* added professional mode
* added member mode switch
* added Portuguese support

astercrm_agentsettings

astercrm agent management
astercrm_clearscreen

astercrm clearscreen
astercrm_dialliatpannel

astercrm diallist pannel
astercrm_panelswitcher

astercrm panels witcher
astercrm_mp3player

astercrm mp3player for recording files
asterbilling_professional

asterbilling professional mode
asterbilling_portuguese

asterbilling portuguese language support
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freepbx2.6 in asterCC BOX 0.13

freepbx2.6 in asterCC BOX 0.13
asternic_realtime

asternic_realtime
asternic_distribution

asternic_distribution


why i cant see booth history when can see the calling call?

By | asterBilling | 2 Comments

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).

live call in booth window

nothing in booth box

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

add callshop & realtime billing feature to your a2billing

By | asterBilling | 2 Comments

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
  • sc-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
  • sc-4
  • 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
  • sc-1
  • login as groupadmin/operator and enjoy 🙂
  • sc-3

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