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msc at freeswitch.org
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PostPosted: Wed Aug 12, 2009 7:43 pm    Post subject: [Freeswitch-users] Question about sharing conference between Reply with quote

On Wed, Aug 12, 2009 at 6:10 PM, Tina Martinez <tina@a2unlimited.com (tina@a2unlimited.com)> wrote:
Quote:
Michael,

Thanks for the welcome, and for the response to my question.

The call control and dynamic setup of conferences I have working (pretty cool stuff).
The tricky part, as you said, is "linking" the servers together.

Basically, what I need to do is establish a connection that will not be dependent
on a live person being on the call.  And I would prefer to avoid having to
register actual phone extensions for every server -- and for every conference call.

I apologize if I'm slow, but I'm new to working an application like this.

No worries. Smile
 

Quote:


In your example, you stated, "3900 is an extension set up in FS_B's dialplan",
does this extension have to be a live person (or soft-phone connected using an
auto-answer mechanism)?  or can I setup something where there is not a phone
actually connected?

No actual telephone is needed. Here's an example dialplan snippet that you could drop right into conf/dialplan/default/ in a new file. (I prefer to put my own custom dialplan entries into a separate file instead of editing default.xml)

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="conference" data="Conf@FS_B"/>
    </condition>
  </extension>
</include>
 
Put the above into conf/dialplan/default/01_ConfB.xml on the "FS_B" server. (You can make a similar file on the "FS_A" server or any other server if you'd like.)

You will also need to create a "public" extension which will route the inbound calls appropriately. (If that doesn't make any sense right now then don't worry, just do it. Smile

Put the following into a file named conf/dialplan/public/01_ConfB.xml:
<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="transfer" data="3900 XML default"/>
    </condition>
  </extension>
</include>

(Again, you can do the same on all of your servers - this will allow all servers to receive calls and route them to x3900.)

Now that you've got 3900 set up you can test it. Press F6 (or type "reloadxml") at the CLI for FS_B. Then, have a phone that is registered to FS_B make a call to 3900. It should be alone in the conference.

Now you'll need to set up some sort of dialplan routing to call from FS_A to FS_B, unless you have a SIP phone registered to FS_A that can dial a SIP URI. The SIP URI is: sip:3900@FS_B.IP.Address. For kicks, let's add a simple dialplan extension on FS_A that allows you to dial "23900" to get to FS_B's 3900 extension. Put this into conf/dialplan/default/01_Dial_ConfB.xml on FS_A:

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^2(3900)$">
      <action application="bridge" data="sofia/internal/$1@FS_B.IP.Address"/>
    </condition>
  </extension>
</include>

Save, and do the reloadxml (or F6) thing on FS_A CLI. Now on FS_A you can dial "23900" and it will ring right into 3900 on FS_B so that the phone registered at FS_A is in the conference on FS_B. Got it?

Have fun tinkering and let us know how it all goes.
-MC



Quote:

Right now I'm able to place a call from one machine to another, but I'm calling
an X-Lite phone on the second server.

Also, if it is possible to call a "virtual" extension, I have no problem
incorporating application logic that would clean-up the orphaned conferences on
all machines when a conference call is complete.  I'm more concerned with being
able to setup the "links" quickly and in an elegant fashion.

- T




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msc at freeswitch.org
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PostPosted: Fri Aug 14, 2009 1:33 pm    Post subject: [Freeswitch-users] Question about sharing conference between Reply with quote

On Fri, Aug 14, 2009 at 11:20 AM, Tina Martinez <tina@a2unlimited.com (tina@a2unlimited.com)> wrote:
Quote:
Michael,

Thanks again for bearing with my novice perspective on this.

I was able to achieve the link between two FS servers as intended.  However, I
was not able to setup a "new" dialplan file as you described.  I had to place the
script into the default.xml dialplan to get it to work.  Is there something I'm
supposed to do to get FS to look in the conf/dialplan/default ?

Look at the sample files that are already in conf/dialplan/default/ and conf/dialplan/public/ to see what they should look like. Just remember to name the files with leading digits, so you have something like this:
conf/dialplan/default/01_My_Custom_Extensions.xml
conf/dialplan/public/01_More_Public_Extensions.xml

The files must be .xml files and preferable should have the <include> and </include> tags at the beginning and end of the files. The other thing you can do is just make a copy of one of the files that's already in conf/dialplan/default/ or conf/dialplan/public/ and edit it.
 

Quote:


Also, in using the code as we discussed, I was also able to establish the link
between the two servers without having an actual soft-phone registered/connected
to the FS server (which is ideal for my situation).  I can create the link
between the servers, and then dial out to external phone numbers from the
respective servers and connect the participants.  Then, regardless of who
hangs-up, the conference between the servers remains in tact.  So far it works
very well.

Glad to hear it!
-MC
 

Quote:

- T

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