BBB configuration to use SIP trunking to make outbound calls

[Rajeev Khurana]

Hi,

I posted this question on users group yesterday. I should have posted it on dev group.

 Has anyone configured BBB to make outbound PSTN call to user's phone? I found discussions that show how to configure DIDs to allow users to dial-in to the conference. My scenario is opposite. I want BBB to provide an option to dial the user. So, when a user joins the call, prompt him/her to put a PSTN number and then BBB would dial the user on that number. When user answers, ask user to press "1" to get connected to the conference. "1" is for  voicemail avoidance. This is similar to WebEx/GoTo Meetings feature where system out dials to the users.

It will be very helpful If anyone has specific example on freeswitch and bbb configuration to achieve this and they can share.

Thanks

-Rajeev

[HostBBB.com]

Rajeev, 

 

1) Once you get a telephone provider,  you need to change your BBB server to listen the external interface.

http://code.google.com/p/bigbluebutton/wiki/FAQ#How_do_I_configure_FreeSWITCH_to_use_127.0.0.1?

This shows the files that need to be edited.

 

2) Then you need to configure a gateway in freeswitch.  www.freeswitch.org has plenty of examples, you need to get the authentication info from your provider, and create a file that is added in sip_profiles.

 

Here is a sample of a asterisk gateway I use, that allows BBB to call out to an asterisk box connected to PSTN.

<include>
  <gateway name="asterisk">
  <param name="username" value="xxxxxxxxx"/>
  <param name="realm" value="xx.xx.xx.xx"/>
  <param name="password" value="bubba_love_bbb"/>
  <param name="register" value="true"/>
  <param name="proxy" value="xx.xx.xx.xx"/>
  <!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
  <!--<param name="register-proxy" value="mysbc.com"/>-->
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <!--<param name="expire-seconds" value="60"/>-->
  <!--/// do not register ///-->
  <!--<param name="register" value="false"/>-->
  <!-- which transport to use for register -->
  <param name="register-transport" value="udp"/>
  <!--How many seconds before a retry when a failure or timeout occurs -->
  <!--<param name="retry-seconds" value="30"/>-->
  <!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
  <!--<param name="caller-id-in-from" value="false"/>-->
  <!--extra sip params to send in the contact-->
  <!--<param name="contact-params" value="tport=tcp"/>-->
  <!--send an options ping every x seconds, failure will unregister and/or mark it down-->
  <!--<param name="ping" value="25"/>-->
  </gateway>
</include>

 

With the gateway configured, you should be able to see BBB register to your provider when you watch fs_cli.  Don't forget that SIP needs some ports open 5060, and the RTP ports for audio.

 

3) To Get outbound calling working

If you join a BBB meeting you will get the voicebridge number assigned.

To test in fs_cli  "conference voicebridge#   dial sofia/profilename/$calln...@outbound.yourprovider.net"

You should see "Outbound Call" in bbb users window and hear it dialing the user, when they answer, they will be connected to meetings.

There audio is recorded.

 

4) Now for the real customization there are a couple of approaches depending on requirements.

 

a) Use an external script that sets up call and does not require any BBB customization. It simply calls the person and adds them to conference, does not unify the users in window. If your front end system is smart enough

b) Pass the number to freeswitch from within bbb, ( I do it through chat module, though not "best design" it works)

c) add new bbb module that has dialer/digitpad and prompts users similar to audiocheck does to get number, dial, and unify.

d) you can also simply create a dialplan option, you dial freeswitch, hit an extension and it prompts you for conference add and number to call.

 

Hope this helps.  1-3 are pretty straight forward.   4 is a function of requirements.

 

If you run into any problems post back into the forums be glad to assist.

 

Regards

Stephen

hostbbb.com