SIP trunking with freeswitch

[Stéphane]

Hi,

I am trying to connect bbb with our sip trunk. It seems, that freeswitch only listens to  127.0.0.1:5060 and not to the external ip. Doesn't freeswitch bind automatically to the external ip? Do I have to enable it?

Regards,

Stéphane

[Fred Dixon]

Hi Stéphane,

We modified the setup of BigBlueButton so FreeSWITCH now binds to 127.0.0.1.  See

    http://code.google.com/p/bigbluebutton/issues/detail?id=1133 

You'll need to modify the FreeSWITCH configuration to listen to an external IP (the above issue shows the files we changed to bind to 127.0.0.1).

Regards,... Fred

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[Stéphane]

Hi Fred,

thanks. I have managed to connect the sip trunk to fs. How do I set up the dial plan in order to connect to bbb conference?

Regards,

Stéphane

Am Montag, 5. März 2012 20:05:25 UTC+1 schrieb Fred Dixon:

Hi Stéphane,

We modified the setup of BigBlueButton so FreeSWITCH now binds to 127.0.0.1.  See

    http://code.google.com/p/bigbluebutton/issues/detail?id=1133 

You'll need to modify the FreeSWITCH configuration to listen to an external IP (the above issue shows the files we changed to bind to 127.0.0.1).

Regards,... Fred

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On Mon, Mar 5, 2012 at 2:02 PM, Stéphane <step...@lux.io> wrote:

Hi,

I am trying to connect bbb with our sip trunk. It seems, that freeswitch only listens to  127.0.0.1:5060 and not to the external ip. Doesn't freeswitch bind automatically to the external ip? Do I have to enable it?

Regards,

Stéphane

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[HostBBB.com]

Stephane, 

Each profile uses a separate listening port.  One way is to display the conference number in welcome message, then have user be prompted for the roomid id, then a security prompt, in the case below the security pin is first 3 digits of the room number + 11.   but can be anything, or nothing at all.

This is what i use.... prompts user twice, first is room, second is security pin

Add %%CONFNUM%% to bigbluebutton.properties defaultWelcomeMessage so users see #

PUBLIC.XML - Add this new extension to public context

<extension name="phone pin">

<condition field="destination_number" expression=“8882299756">

<action application="answer"/>

<action application="sleep" data="500"/>

<action application="play_and_get_digits" data="2 5 3 7000 # conference/conf-pin.wav ivr/

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>

ivr-that_was_an_invalid_entry.wav pin \d+"/>

<action application="transfer" data="SEND_TO_CONFERENCE XML default"/>

</condition>

</extension>

PIN IMPORTANT, this one uses first 3 digits of conf, and then 11

Regards,

Stephen

http://hostbbb.com

[Stéphane]

Hi Stephen,

thank you very much. It works perfectly.

What do I have to change, in order to have the user only enter the room id, without the second security pin?

Regards,

Stéphane

[HostBBB.com]

Change  <action application="conference" data="${pin}@wideband+${pin:0:3}11"/> 

to   <action application="conference" data="${pin}@wideband"/> 

And that should do it, however any one can call your switch and enter a valid conf number to get a empty conf room, than tell their buddies to call in with no security,  you wont even know there using your system.  Also make sure outbound toll calling is not setup thru your switch....  dont want a big phone bill.

Its also just as easy to connect skype clients and a HD G.722 clients to the conference as well.

regards.

Stephen

http://hostbbb.com

[brunoBR]

Hey guys!

I'm having a similar issue, after upgrading to beta 0.4 my current
office PBX was unable to dial into the BBB voice conference using SIP
trunking. When I changed the listening IP on vars.xml from 127.0.0.1
to the LAN address of the server, I was able to dial into the
conference call, but then web clients now can't join the voice
conference.

I do get an error on red5 when doing bbb-conf --clean, which is " *
Starting Red5 Server red5 ...... Error: FreeSWITCH SIP 127.0.0.1
didn't start". BBB still manages to go up though.

Is there a way to get back to the previous voice conference settings
on beta 0.3? I know this is supposed to be an improvement on security,
but I'm running BBB on a private LAN, its not opened on the internet,
and it made it difficult to connect other SIP capable devices to BBB.

Thanks!

[Fred Dixon]

Hi BrunoBR,

The changes we made to the FreeSWITCH configuration (along with how to rever to the previous settings in 0.8-beta-3) are documented in this issue

  http://code.google.com/p/bigbluebutton/issues/detail?id=1133

Regards,... Fred

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[Bruno Amaral]

Thanks Fred! I managed to make it work!

[v0id]

[Tommy Jack]

Hello Stephane,

I am trying to Join BBB conference from my Elastix Client.

I can manage to call FreeSwitch and I can hear the voice "You are the only one in this conference"

Now i have two major problem:

1) i couldn't see the VoIP user in any BBB conference room. VoIP users can talk each other.

2) I want to put PIN number for the VoIP users to join into the conference.

Below is the current public.xml and default.xml.
I tried to put your config but when I reload BBB, the free couldn't start.

I think something wrong in this config. I am really sorry, i am very new in programming.

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>

ivr-that_was_an_invalid_entry.wav pin \d+"/>

<action application="transfer" data="SEND_TO_CONFERENCE XML default"/>

</condition>

</extension>

Current defaut.xml

<include>

  <context name="default">

    <extension name="unloop">

      <condition field="${unroll_loops}" expression="^true$"/>

      <condition field="${sip_looped_call}" expression="^true$">

        <action application="deflect" data="${destination_number}"/>

      </condition>

    </extension>

    <!--

        You can place files in the default directory to get included.

    -->

    <X-PRE-PROCESS cmd="include" data="default/*.xml"/>

And public.xml

<include>

  <context name="public">

    <extension name="unloop">

      <condition field="${unroll_loops}" expression="^true$"/>

      <condition field="${sip_looped_call}" expression="^true$">

<action application="deflect" data="${destination_number}"/>

      </condition>

    </extension>

    <!--

Tag anything pass thru here as an outside_call so you can make sure not

to create any routing loops based on the conditions that it came from 

the outside of the switch.  

    -->

    <extension name="outside_call" continue="true">

      <condition>

<action application="set" data="outside_call=true"/>

<action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>

      </condition>

    </extension>

    <!--

        You can place files in the public directory to get included.

    -->

    <X-PRE-PROCESS cmd="include" data="public/*.xml"/>

    

  </context>

</include>

Your help is highly appreciated.

Respectfully,

Tommy

[HostBBB.com]

Tommy, you need to add to the dialplan the ability to prompt for a voice bridge to route to the actual conference,  what is happening now, it you are being placed into some adhoc freeswitch conf all by your self.  So wont show in BBB.

add to bigbluebutton.properties  welcome message PIN: %%CONFNUM%% and you will see the actual voicebridge number for the BBB session,

then you need to add prompt in dialplan to ask for this number, and route to the correct conference.  Not sure if your using .81 or .90beta.

but here is the code I use to match and prompt for pin in public context.

<include>
    <extension name="bbb_did_call" continue="true">
      <condition field="destination_number" expression="^yournumber$" break="on-false">
        <action application="set" data="bbb_authorized=true"/>
<action application="answer"/>
    <action application="sleep" data="1500"/>
    <action application="play_and_get_digits" data="4 5 3 6000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+ 2000 'NO_PIN XML default'"/>
    <action application="log" data="INFO The value of pin ${pin}"/>

    <action application="transfer" data="SEND_TO_CONFERENCE XML default"/>

        <action application="transfer" data="${destination_number} XML default"/>
      </condition>
    </extension>
</include>

also it you connect now.

sudo /opt/freeswitch/bin/fs_cli

>> conference list

this will actually show you the name of adhoc conference you currently connect to.

regards,

Stephen

hostbbb.com

[Tommy Jack]

Hello Stephen, 

Thank you very much for your reply.

I have put the code and it works beautifully now.

The only problem is if the user dial the wrong PIN, it goes to the random conference room.

I want the user not to go any conference if the PIN (actual conference voiceBridge> is wrong.

Would you mind to help and i could finish my project.

Thank you in advance.

with regards,

Tommy

[HostBBB.com]

Tommy, if you add the condition to dialplan match.

<condition field="${conference ${pin} list}" expression="/sofia/g" />

<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

do stuff here,

</condition>

This will return true only if the conference is active.  Meaning, at Least 1 user needs to already be in conf, to get in.

If not you simply play a sound prompt that "Meeting Has not started".  this stops someone from fat fingering a number and sitting in the wrong conference room.

Its also possible to prompt user for their name, so freeswitch announces them in the conference.   the freeswitch wiki has lots of good examples.

regards,

Stephen

hostbbb.com

[Tommy Jack]

Hello Stephen,

Fantastic.!!!!!!!!!!

But one more help please. Now the call is totally drop without notifying "Invalid PIN Number" if it is wrong.

I want user to retry 3 the correct  PIN for three times with notification.

Below is my current config.

Really appreciate your kind help.

PUBLIC.XML

   <extension name="phone pin"> 

     <condition field="destination_number" expression="71111"> 

      <action application="answer"/> 

      <action application="sleep" data="500"/> 

      <action application="play_and_get_digits" data="5 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>

DEFAULT.XML

 <extension name="phone conference"> 

      <condition field="${conference ${pin} list}" expression="/sofia/g" />

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$"> 

       <action application="conference" data="${pin}@wideband"/> 

      </condition> 

    </extension>

[Tommy Jack]

Hello Stephen,

I am still struggling to put the "Invalid pin number reply".

That could be complete my project.

On Tuesday, 6 March 2012 03:02:04 UTC+8, Stéphane wrote:

[HostBBB.com]

Tommy, you can do some advanced dialplan logic and count the attempts looping back thru dialplan using loopback but the simplest thing to do is add right below your current logic this extension.

When it fails to enter "phone conference" above, it will match this next one.   Change the audio wav file to whatever you want.

<extension name="conference_not_active_or_no_valid_pin_entered">

<condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$"> 

   <action application="playback" data="conference/conf-goodbye.wav"/>
   <action application="hangup"/>
</condition>
</extension>

You now will not hear the dropped call.

regards,

Stephen

hostbbb.com

[Tommy Jack]

Hello Stephen,

Perfect. And very appreciate for your helps.

I have done the project and put in the production now.

I post my dial-plan below for anyone who needs.

I am not good in programming so everyone who has a better option, please kindly suggest.

Thank you everybody.

Asterisk-FreeSwitch Dial-Plan

PSTN=>ELX=>FreePBX (MD)=>BBB-Freeswitch=>BBB-CONF

VoIP=>ELX=fs-gw(Trunk)=> BBB-Freeswitch=>BBB-CONF

/opt/freeswtich/conf/dialplan/public.xml/Public Context

<extension name="phone pin">

     <condition field="destination_number" expression="71111">

      <action application="answer"/>

      <action application="sleep" data="500"/>

      <action application="play_and_get_digits" data="5 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>

/opt/freeswtich/conf/dialplan/public.xml/default Context

<extension name="phone conference">

      <condition field="${conference ${pin} list}" expression="/sofia/g" />   

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

       <action application="conference" data="${pin}@wideband"/>

      </condition>

    </extension>

 

    <extension name="conference_not_active_or_no_valid_pin_entered">

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

        <action application="play_and_get_digits" data="5 5 2 5000 #

         conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>

        <action application="transfer" data="SEND_TO_CONFERENCE XML dropme1"/>         

      </condition>

    </extension>

 /opt/freeswtich/conf/dialplan/public.xml/dropme1 Context

<context name="dropme1">

    <extension name="phone conf1">

      <condition field="${conference ${pin} list}" expression="/sofia/g" />  

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

       <action application="conference" data="${pin}@wideband"/>

      </condition>

    </extension>

 

    <extension name="conference_not_active_or_no_valid_pin_entered1">

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

        <action application="play_and_get_digits" data="5 5 1 5000 #

         conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>

        <action application="transfer" data="SEND_TO_CONFERENCE XML dropme2"/>  

<context name="dropme2">

    <extension name="phone conf2">

      <condition field="${conference ${pin} list}" expression="/sofia/g" />  

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

       <action application="conference" data="${pin}@wideband"/>

      </condition>

    </extension>

 

    <extension name="conference_not_active_or_no_valid_pin_entered1">

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

        <action application="answer"/>

        <action application="sleep" data="500"/>

        <action application="play_and_get_digits" data="5 5 1 5000 #

         conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>

        <action application="transfer" data="SEND_TO_CONFERENCE XML dropme3"/>        

      </condition>

       <context name="dropme3">

    <extension name="phone conf3">

      <condition field="${conference ${pin} list}" expression="/sofia/g" />  

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

       <action application="conference" data="${pin}@wideband"/>

      </condition>

    </extension>

 

    <extension name="conference_not_active_or_no_valid_pin_entered3">

      <condition field="destination_number" expression="^(SEND_TO_CONFERENCE)$">

        <action application="playback" data="conference/conf-goodbye.wav"/>

        <action application="hangup"/>        

      </condition>

    </extension>

  </context>

[Tommy Jack]

Hello Stephen,

Obviously, The dial-plan is very long and complicated.

Can we put the condition like
If pin number (not equal) to active BBBVoiceBridge then reply "Invalid Pin Number"

If retry fails for 3 times, then say good bye.

Above dial-plan does it but i don't this it is very professional in terms of programming.

Thank you very much Stephen.

with regards,

Tommy

On Tuesday, 6 March 2012 03:02:04 UTC+8, Stéphane wrote:

[Paul Belcher]

I am new to programing in this way too.  I have though a question, which directory did you place your dropme1.xml files and does each context block have to be a separate file or can they be nested in an "Include" block and put all the same file?  Thanks for whatever information you can provide me.  Paul

[Nemanja Manic]

Hello,

Is it the same process if I want to connect using Ekiga for example? I should open up FreeSWITCH as explained here: https://code.google.com/p/bigbluebutton/issues/detail?id=1133 ?

I tried to understand it but no luck so far.

[Richard Alam]

On Fri, Jan 15, 2016 at 1:59 PM, Nemanja Manic <neman...@gmail.com> wrote:

Hello,

Is it the same process if I want to connect using Ekiga for example? I should open up FreeSWITCH as explained here: https://code.google.com/p/bigbluebutton/issues/detail?id=1133 ?

I tried to understand it but no luck so far.

You edit /opt/freeswitch/conf/vars.xml to listen using the external ip.

  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.23.19"/>

  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

In /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties

bbb.sip.app.ip=192.168.23.19

bbb.sip.app.port=5070

# The username and password the BBB SIP app to use to

# register with FreeSWITCH

sip.server.username=bbbuser

sip.server.password=secret

# The ip and port of the FreeSWITCH server

freeswitch.ip=192.168.23.19

freeswitch.port=5060

Restart bigbluebutton. Then on your Ekiga, you configure it to connect to <IP>:5060

Richard

 

On Monday, 5 March 2012 20:02:04 UTC+1, Stéphane wrote:

Hi,

I am trying to connect bbb with our sip trunk. It seems, that freeswitch only listens to  127.0.0.1:5060 and not to the external ip. Doesn't freeswitch bind automatically to the external ip? Do I have to enable it?

Regards,

Stéphane

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[Nemanja Manic]

Hello Richard,
Thank you for your response.

I tried as you suggested but it's still not working.
I replaced your 192.168.23.19 with my server IP address, of course, but when I try to enter aaa.bbb.ccc.ddd:5060 and call from Ekiga, it just says "Could not connect to remote host".

Shouldn't I try calling some...@192.168.23.19 or something similar?

I need to apologize for my ignorance, I'm still studying and really struggling to understand everything about SIP.

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[Richard Alam]

On Fri, Jan 15, 2016 at 5:24 PM, Nemanja Manic <neman...@gmail.com> wrote:

Hello Richard,
Thank you for your response.

I tried as you suggested but it's still not working.
I replaced your 192.168.23.19 with my server IP address, of course, but when I try to enter aaa.bbb.ccc.ddd:5060 and call from Ekiga, it just says "Could not connect to remote host".

Are you sure that there isn't any firewall blocking the ports (tcp/udp) between your Ekiga and FS?

 

Shouldn't I try calling some...@192.168.23.19 or something similar?

I haven't used Ekiga so I don't know how to make calls with it. I've use Zoiper before. I just call the conference number using the dialpad after configuring Zoiper to connect to FS.

Are there any logs on Ekiga? Maybe a sip trace?

Can you try to telnet to <FSIP>:5060 from your Ekiga machine?

Richard

 

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[HostBBB.com]

quick test to see if freeswitch is listening on external and not blocked by firewalls is to 

telnet yourserver.com 5060  from a remote system.

If you get a connection everything is ready to go.

also once you get their,

the dialplan needs to see the call and match some regex to get you to the bbb conference with the right vbridge

sudo /opt/freeswitch/bin/fs_cli  will show the console.   You should see the console display a lot of lines when hitting the dialplan,

there are some other posts in forums, and in this earlier thread on how to add the dialplan logic to prompt for voicebridge.

regards.

Stephen

hostbbb.com

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[Nemanja Manic]

First of all, thank you all for your patience.
I did some reading for a few hours now, and I'm starting to get an idea how everything works.
I'm not bound to any particular software, I'm just trying to make it somehow work. So I've tried X-Lite, Zoiper, , Jitsi and Ekiga so far. I managed to register users only with X-Lite and Zoiper, but that's enough for me for testing.

This is what I did so far:
I managed to register 2 users.
I successfully made some simple extension to playback some audio to the users dialing a number.

I added the code that HostBBB has given before to the Public and Default context and now when I dial 8882299756 I'm asked about PIN.
When I enter the PIN nothing happens.
This is the log from fs_cli I think is relevant:

2016-01-16 03:10:26.925557 [NOTICE] switch_ivr.c:1863 Transfer sofia/external/10...@1xx.xx.xx.xx to XML[SEND_TO_CONFERENCE@default]

2016-01-16 03:10:26.925557 [INFO] mod_dialplan_xml.c:635 Processing 10001 <10001>->SEND_TO_CONFERENCE in context default

2016-01-16 03:10:26.925557 [INFO] switch_core_state_machine.c:241 No Route, Aborting

2016-01-16 03:10:26.925557 [NOTICE] switch_core_state_machine.c:242 Hangup sofia/external/10...@1xx.xx.xx.xx [CS_ROUTING] [NO_ROUTE_DESTINATION]

2016-01-16 03:10:26.925557 [NOTICE] switch_core_session.c:1641 Session 12 (sofia/external/10...@1xx.xx.xx.xx) Ended

2016-01-16 03:10:26.925557 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/10...@1xx.xx.xx.xx [CS_DESTROY]

I guess that this "NO_ROUTE_DESTINATION" part is a problem, but don't know how to fix it.
Any ideas what it might be?

[Nemanja Manic]

Ok, small update.

I have changed the expression so that it doesn't require +11 for the PIN (as HostBBB.com explained in the message from 08/03/2012) and successfully joined the conferencing by dialing 8882299756 and then entering full 5 digit CONFNUM.
Now everything works as it should.

Again, thank you so much for your help.

[Sujatha Alluru]

I am facing issues with setting the external IP to freeswitch server configuration

I have set the external_ip to the following files

1. /opt/freeswitch/conf/vars.xml
1. edited the ip address with my external address for the following as per the instructions

1.  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.23.19"/>
2.   <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>

1. /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties
   
1. edited the following with my external ip address too
2. bbb.sip.app.ip=192.168.23.19
   
3. freeswitch.ip=192.168.23.19
   

Did a clean start of bbb using 

sudo bbb-conf --clean

When a ran the bbb-conf --check, I got the following warning..

 Warning: The setting of (54.174.182.155) for bbb.sip.app.ip in

#

#    /usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties

#

# does not match the local IP address (172.31.60.119).

# (This is OK if you've manually changed the values to an external

# FreeSWITCH server.)

# Error: Could not detect FreeSWITCH listening on port 5060

I tried to check the service running on port 5060 using " netstat -ant | grep 5060"

tcp6       0      0 ::1:5060                :::*                    LISTEN

Am I missing some other configuration ?  Please help.

Thanks

Sujatha

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