How to VoIP call a meeting room?

[f.erfurth]

Hi, we're developing a software. Now we're trying to get audio-connection. My colleague want use VoIP. Does it mean he has to connect to Asterisk directly? If so, is there any documentation? How can we connect the right VoIP-Channel of a meeting?

 

cu Floh

[HostBBB.com]

f.erfurth... asterisk and freewitch both can be configured to accept
calls from standard sip clients... You just need to add extensions to
the dialplan and dial into the active confernece. We use xlite and
linphone successfully to do this with freeswitch.

To configure voip,... just need to read the asterisk documentation for
adding a sip client.

Regards,
Stephen
http://hostbbb.com

[f.erfurth]

Hi Stephen,

thank you for your quick reply. We're using asterisk here.

 

How can I know which call goes to which conference? I have a /etc/asterisk/bbb_sip.conf which contains following:

>>>

[3000]
type=friend
username=3000
insecure=very
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=bbb-voip
disallow=all
allow=ulaw
<<<

 

It repeats till 3029. So I have 30 "numbers" for VoIP. But I didn't found out yet how BBB or Asterisk decided to which conference a call should go.

 

cu Floh

[Damian]

Hi Floh,

I'm in a similar situation, using BBB with Freeswitch. As far as I
can tell, it assigns each conference a random 5 digit extension number
starting with a "7". I can watch the output of fs_cli to see what
this extension is, and can successfully transfer calls from our VoIP
system to these conference numbers, but that doesn't help my users
much.

This area of BBB seems to be somewhat lacking in documentation. I see
there is a voiceBridge parameter for the create API, which sounds like
it lets you specify the extension number to use. I don't see any way
to get the current number for a conference though. Ideally, there
would be some option to have it display this number in the default web
interface. Maybe there is, and I'm missing it?

- Damian

[Richard Alam]

Hi,

bbb-sip.conf isn't used anymore in v0.71a...look into
bbb_extensions.conf on how the call is handled to put
the user into the conference.

Richaerd

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[Richard Alam]

Hi,

On Mon, Sep 5, 2011 at 2:12 PM, Damian <damian....@gmail.com> wrote:
> Hi Floh,
>
> I'm in a similar situation, using BBB with Freeswitch.  As far as I
> can tell, it assigns each conference a random 5 digit extension number
> starting with a "7".  I can watch the output of fs_cli to see what
> this extension is, and can successfully transfer calls from our VoIP
> system to these conference numbers, but that doesn't help my users
> much.

You can change /opt/freeswitch/conf/dialplan/public.xml to whatever
pattern you want for conference nums.

Here is says any 5-digit number.

<extension name="bbb_conferences">
<condition field="destination_number" expression="^(\d{5})$">
<action application="answer"/>
<action application="conference" data="$1@wideband"/>
<!-- <action application="conference" data="$1@wideband"/> -->
</condition>
</extension>

>
> This area of BBB seems to be somewhat lacking in documentation.  I see
> there is a voiceBridge parameter for the create API, which sounds like
> it lets you specify the extension number to use.  I don't see any way
> to get the current number for a conference though.  Ideally, there

You can try calling the getMeetingInfo API to get the conference bridge.

richard

[Fred Dixon]

Hi Damian,

You can use the welcomeMsg (passed as a parameter to the 'create' API)
to output the current conference number to the user in the chat
welcome message. See

http://code.google.com/p/bigbluebutton/wiki/API#Create_Meeting_(create)

In short, put a "%%CONFNUM%%" in your welcome message and
BigBlueButton will replace it with the voice bridge number for the
current session.

Regards,... Fred

P.S. We'll in the process of updating the documentation for the
upcoming 0.8-beta and will be adding more information around the use
of voiceBridge.

--
http://code.google.com/p/bigbluebutton/wiki/FAQ#BigBlueButton_Committer

On Mon, Sep 5, 2011 at 2:12 PM, Damian <damian....@gmail.com> wrote:

[f.erfurth]

Hi Richard & Fred,
thank you very much. I'll try it out tomorrow. VoiceBridge is available here.

cu Floh

Am 05.09.2011 21:04 schrieb "Fred Dixon" <ffd...@gmail.com>:

[Damian]

On Sep 5, 3:01 pm, Fred Dixon <ffdi...@gmail.com> wrote:
> In short, put a "%%CONFNUM%%" in your welcome message and
> BigBlueButton will replace it with the voice bridge number for the
> current session.

Brilliant. I knew the option was there somewhere.

Many thanks,
- Damian

[HostBBB.com]

in .71a can't yet get the voicebridge in api from existing meeting,
issue 844 (Add <voicebridge> to GetMeetingInfo)

but you can create a meeting passing in the voicebridge, so your
business logic knows how to connect from the start.

Also, to add a pin in freeswitch, and send your inbound dids and sip
phones to it.

DEFAULT.XML - Add this line in default context
<extension name="phone conference">
<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)
$">
<action application="conference" data="${pin}@wideband+${pin:0:3}11"/>
</condition>
</extension>

This makes the password pin first 3 digits of conference + 11 so
room 74523 would have unique pin 74511

regards,
Stephen
http://hostbbb.com

On Sep 5, 2:20 pm, Richard Alam <ritza...@gmail.com> wrote:
> Hi,
>

[f.erfurth]

Hi,

I tried to call the room. I tried on freshly installed bbb server 0.71a. After creating a room with voiceBridge=20000 passed as parameter I tried to call. This is SIP-Adress I tried:

sip:20...@172.19.85.24

 

But linphone just hangup after I pressed "connect". On Server I looked into /var/log/asterisk/messages:

 

[Sep  8 19:02:49] NOTICE[3886] chan_sip.c: Call from '' to extension '20000' rejected because extension not found.

 

So... what did I wrong here?

 

cu Floh

[Richard Alam]

On Thu, Sep 8, 2011 at 1:07 PM, f.erfurth <f.er...@googlemail.com> wrote:
> Hi,
> I tried to call the room. I tried on freshly installed bbb server 0.71a.
> After creating a room with voiceBridge=20000 passed as parameter I tried to
> call. This is SIP-Adress I tried:
> sip:20...@172.19.85.24
>

> But linphone just hangup after I pressed "connect". On Server I looked into
> /var/log/asterisk/messages:
>
> [Sep  8 19:02:49] NOTICE[3886] chan_sip.c: Call from '' to extension '20000'
> rejected because extension not found.
>

You Asterisk dialplan isn't setup properly. Make sure that the call is
forwarded to the right context and
there is an extension that matches 20000

Richard

> So... what did I wrong here?
>
> cu Floh
>
> On Mon, Sep 5, 2011 at 11:02 PM, f.erfurth <f.er...@googlemail.com> wrote:
>>
>> Hi Richard & Fred,
>> thank you very much. I'll try it out tomorrow. VoiceBridge is available
>> here.
>>
>> cu Floh
>>
>> Am 05.09.2011 21:04 schrieb "Fred Dixon" <ffd...@gmail.com>:
>

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[f.erfurth]

Hi,

On Thu, Sep 8, 2011 at 7:11 PM, Richard Alam <ritz...@gmail.com> wrote:

You Asterisk dialplan isn't setup properly. Make sure that the call is
forwarded to the right context and
there is an extension that matches 20000

 

I thought there is already extension installed by bigbluebutton as following file.

 

/etc/asterisk/bbb_extensions.conf

; BigBlueButton extensions.
; In your /etc/asterisk/extensions.conf, add the following line at the end
; of the file
; #include "bbb_extensions.conf"
;
; BigBlueButton: Setup sample conference
[bigbluebutton]
exten => _.,1,Goto(start-dialplan,s,1)
exten => _.,n,Hangup

[start-dialplan]
exten => s,1,Set(TRIES=1)
exten => s,n,Wait(2)
exten => s,n,Answer
exten => s,n,Goto(prompt,s,1)

[prompt]
exten => s,1,Read(CONF_NUM,conf-getconfno,6,,3,10)
exten => s,n,Goto(bbb-conference,${CONF_NUM},1)

; No need to check if conference is valid as they won't be able to login
; if the conference is invalid.
;
[bbb-voip]
exten => _XXXXX.,1,Playback(conf-placeintoconf)
; exten => _XXXX.,n,MeetMe(${EXTEN},cdMsT)
exten => _XXXXX.,n,Konference(${EXTEN},H)

 

[bbb-conference]
include => echo-test

exten => _XXXXX.,1,Agi(agi://localhost/findConference?conference=${EXTEN})
exten => _XXXXX.,n,GotoIf($[${EXTEN} = ${CONFERENCE_FOUND}]?valid:invalid)
exten => _XXXXX.,n(valid),Playback(conf-placeintoconf)
; exten => _XXXX.,n,MeetMe(${CONFERENCE_FOUND},cdMsT)
exten => _XXXXX.,n,Konference(${CONFERENCE_FOUND},H)
exten => _XXXXX.,n(invalid),Goto(handle-invalid-conference,s,1)

[handle-invalid-conference]
exten => s,1,Playback(conf-invalid)
exten => s,n,GotoIf($[${TRIES} < 3]?try-again:do-not-try-again)
exten => s,n(try-again),Set(TRIES=$[${TRIES} + 1])
exten => s,n,Goto(prompt,s,1)
exten => s,n(do-not-try-again),Hangup

[echo-test]
;
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Answer                   ; Do the echo test
exten => 600,n,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo                     ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)                ; Start over

 

cu Floh

[f.erfurth]

Hi, I'm still looking for solution why I get error ala no such extension 20000. I tried to understand asterisk-documentation. I just want to be sure:

 

Before the extension file contained strings like XXXX. I added a X so there are XXXXX now. The X means digits, am I right?

 

cu Floh

[Richard Alam]

Hi,

What context are you putting when calling 2000?

I assume you register your linphone with Asterisk. So the account you
tie your linphone will have a context entry.
That context should be able to access [bigbluebutton] or [bbb-voip] context.

See http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf

Richard

[f.erfurth]

Hi Richard,

On Mon, Sep 12, 2011 at 3:02 PM, Richard Alam <ritz...@gmail.com> wrote:

What context are you putting when calling 2000?

 

Huh? Sorry, but I didn't understand what you mean.

 

I assume you register your linphone with Asterisk. So the account you
tie your linphone will have a context entry.

 

So I need an asterisk account?

 

See http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf

 

Thanx. I'll read this and try it out. Sounds interesting...

 

cu Floh