What do I need to set up a bridge for my users to dial into if they don't want to join through web?

[Aleksandra Czajka]

Hi All,

Sorry if this has been asked before, but, I can't find one place to find a succinct answer and steps. Seems like this should be readily available. 

All I need to to is provide my participants with a phone number and bridge number to call in. I know I can set the bridge number with the create meeting call and I know it needs to be 5 digits and start with a 7. I've done that. However, I'm not sure what to do next. I try calling the default number that is provided in bigbluebutton.properties for tomcat6 and it doesn't dial. Just drops on me.

Do I need to set up anything else for the default number to work? It doesn't state what to do anywhere that I've seen.

Your help greatly appreciated in advance.

Best,

Aleks

[Calvin Walton]

Hi,

Unfortunately, there is no free incoming calling available for
BigBlueButton; the number in the configuration file is just an example,
not a real number.

The bridge number is required even if you do not allow incoming phone
calls, since it's used internally BigBlueButton to create the conference
in FreeSWITCH that will handle the voice of users who connect via the
Flash interface.

In order to get incoming calls to work with your BigBlueButton server,
you will have to contact a VOIP provider, and purchase a plan that will
include a DID (basically, a phone number). You can then reconfigure
FreeSWITCH on the BigBlueButton server to register with your VOIP
provider and accept incoming calls.

This can get pretty complicated to set up, and I don't think we have any
detailed docs for it. Feel free to ask for further help, or you could
try contacting one of the commercial support companies:
http://www.bigbluebutton.org/commercial-support/
as many of them have experience integrating BigBlueButton with VOIP
providers.

--
Calvin Walton <calvin...@kepstin.ca>
BigBlueButton Developer

[Aleksandra Czajka]

:'-(

Ok, thank you for your help! I think I'll just go without it for now.

Best,

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Aleksandra Czajka

Founder, Coach Base, Video Interview Practice

https://CoachBase.org

https://VideoInterviewPractice.com

http://www.linkedin.com/pub/aleksandra-czajka/5/3a4/13b

[abdul waheed]

 """""""""""""""""In order to get incoming calls to work with your BigBlueButton server,
you will have to contact a VOIP provider, and purchase a plan that will
include a DID (basically, a phone number). You can then reconfigure
FreeSWITCH on the BigBlueButton server to register with your VOIP
provider and accept incoming calls  """""""""""""""""""""""""

Can you please provide step by step instructions on how to reconfigure FreeSWITCH on the BigBlueButton server to register with your VOIP provider and accept incoming calls.

[Gene Schenk]

Hello,
John at  Covici Computer Systems may be
able/willing to help.  He helped me to set up the
FreeSwitch config and voip system at a very
reasonable price.

Send him an email:  covici at css dot covici dot com
Good luck,
Gene

[HostBBB.com]

Abdul, here are the general steps

 

1) Find ITSP provider that has local numbers for you callers and provided IP authentication.

2) Determine the number of concurrent callers you need and get sip trunks to support. ( A cheap DID normally only supports 2 channels)

3) re-configure your Freeswitch to listen on external interface (currently 127.0.0.1) and open port 5060 to world.

4) to prevent being hacked to death, add iptables rules to only accept traffic from your ISTP.

5) Edit the dialplans in freeswitch to accept inbound calls from this number, and prompt for conference number and pin.

6) Add DIALNUM%% and %%CONFNUM%% to welcome param at create or default bigbluebutton.properties so voicebridge appears for users.

There are few providers on commercial support list that can provide part of all of this solution if needed.

 

Also,  if you want to try and do it yourself.   read the freeswitch wikis, and books.   they have great examples on how to do this.  Post back in the forums if you get stuck on pieces and we can assist.

 

You can also bridge Skype, and Skype connect directly in BBB for an affordable inbound solution.

 

Regards,

Stephen

hostbbb.com

[Stéphane]

Abdul,

I have posted at https://drupal.org/node/1786614#comment-7689551 how I manged to set up dialing into a conference.

[HostBBB.com]

[HostBBB.com]

that code looks familiar :)

[abdul waheed]

Thank you Stephen,

I tried it myself and need your assistance. I followed the instructions on Freeswitch vikis, now bbb starts but no user shows in listeners when I create meeting from demo pages.

Regarding Skype, I installed Skype module, but I try to start, it gives the following error,

zulu1449:/usr/src/freeswitch/src/mod/endpoints/mod_skypopen/install# sh /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh
ERROR: Module snd_pcm_oss does not exist in /proc/modules
ERROR: Module snd_mixer_oss does not exist in /proc/modules
ERROR: Module snd_seq_oss does not exist in /proc/modules
mknod: `/dev/dsp': File exists
insmod: error inserting '/usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko': -1 File exists
/usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh: 14: /usr/local/freeswitch/skypopen/skype-clients-startup-dir/start_skype_clients.sh: /usr/bin/Xvfb: not found

Can we receive skype calls in BBB sessions with Skype module in Freeswitch?,

Second question is that if we configure Skype Connect with SIP, can we receive regular skype calls, or only receive DID skype calls?


Regards,
Abdul Waheed

[Stéphane]

Credits go to you ;)

[HostBBB.com]

With Skype connect you can create a Skype user  like  waheed-connect,  and  route it to any BBB server.  Can also turnup DIDs and tie it to the same set of sip trunks.   You need to secure your sip port and lock down so only valid sip connections are made and rate limited.   with a little extra coding you can actually originate outbound calls from BBB to students cell phones direct and they join the bbb session.

 

Looks like you are trying to install the Skype modules in freeswitch.   this actually starts up Linux sip clients and is different and more complex route.  I wouldn't use this in production.  Since all indications is its not going to work long term as Microsoft/skpye merge their Lync/Skype offerings and shut down support for APIs.

 

regards,

Stephen

hostbbb.com

[qurant...@gmail.com]

The user I create like waheed.connect, can I receive normal Skype calls from skype users on it, and then rout to BBB sessions?

Best Regards,
Abdul Waheed

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[HostBBB.com]

you tie it to a sip channel,  people call it on Skype and routes somewhere depending the sip settings.  This can be directly to a BBB class in session, or to your office phone if one is not running.

 

With a little logic in freeswitch dialplan you can add Skype names for every tutor and have ability to route and record, or just have one to answer all calls.

 

regards,

Stephen

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[Romeo]

Is it possible to add multiple DID numbers (e.g to have Dial In Numbers in multiple countries) to the FreeSwitch configuration? 

[HostBBB.com]

Romeo,

Yes,  you can add multiple DIDs from around the globe to point to one BBB server,  just need add a dialplan entry for each number in freeswitch

The freeswitch wiki has a lot of good examples and list of global providers.

regards,

Stephen

hostbbbb.com

[Romeo]

Dear Stephen,

Thanks a lot for your feedback!  I think I almost nailed it (using one phone number to start with), however after I enter the conference PIN code the call gets terminated.

This is what I've done so far:

1. Reconfigured Freeswitch to listen on external interface

2. Opened port 5060 TCP and UDP

3. Added the following config in the /opt/freeswitch/conf/dialplan/public.xml (public context):

<extension name="phone pin">
<condition field="destination_number" expression="[YOUR DIALIN NUMBER]">
<action application="answer"/>
<action application="sleep" data="500"/>
<action application="play_and_get_digits" data="2 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
<action application="transfer" data="SEND_TO_CONFERENCE XML default"/>
</condition>
</extension>

4. Added the following config in the file /opt/freeswitch/conf/dialplan/default.xml  (default context):

<extension name="phone conference">
<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)quot;>
<action application="conference" data="${pin}@wideband"/>
</condition>
</extension>  

Any ideas? Many thanks in advance!

[Romeo]

Please disregard my last post, everything works perfectly now.  Initially I added the config parameters to the bottom of both xml files, I now moved them up to the top and that did the trick.

Regarding multiple phone numbers, I suppose I have to add them with multiple conditions, e.g as follows:

<extension name="phone pin">
<condition field="destination_number" expression="phone number 1|phone number 2|phone number 3">

<action application="answer"/>
<action application="sleep" data="500"/>
<action application="play_and_get_digits" data="2 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
<action application="transfer" data="SEND_TO_CONFERENCE XML default"/>
</condition>
</extension>

Thanks for your valuable input!

Kind regards,
Romeo

On Tuesday, January 28, 2014 7:33:24 PM UTC+2, HostBBB.com wrote:

[Romeo]

Just for your info, the OR conditions in the dialing plan seems to work, I've added two numbers as follows:

<extension name="phone pin">
<condition field="destination_number" expression="phone1|phone2">

<action application="answer"/>
<action application="sleep" data="500"/>
<action application="play_and_get_digits" data="2 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
<action application="transfer" data="SEND_TO_CONFERENCE XML default"/>
</condition>
</extension>

I was able to get to the PIN prompt using both numbers.  Great system!