freeswitch why cant I dial local extensions

[Ryan Allen]

Hi Please help! I have had bigbluebutton 0.7 with asterisks and I was
able to configure asterisk as are phone server and bbb as video /
document viewer with no issues. But I like the video feed mod in
bigbluebutton 8. So I installed bbb and configured freeswitch to have
out going calls incoming calls to a IVR. Can call into BBB off the
IVR. Everything is great but.... I am able unable to dial another sip
phone connected to freeswitch. I get errors of 486, 404 or just
unreachable, not found, to many hops. No matter How I configure the
dial plan I get this. I saw that all the phones where connected as
external so I connected them internal using (SERVERIP):5090. Can
anyone help me with this one. I just cant seem to get it.
Thanks,
Ryan

[Fred Dixon]

Hi Ryan,

In BigBlueButton 0.8-beta-4, FreeSWITCH now only listens to the local loopback (127.0.0.1) address.  You can change this to listen to the external IP address (so FreeSWITCH can receive in coming calls).   See

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

Regards,... Fred

-- 
BigBlueButton Developer

http://bigbluebutton.org/
http://code.google.com/p/bigbluebutton

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[Ryan Allen]

I did this. It hears everything I can get there from my external ip
just fine. My sip phones register to freeswitch just fine. My issue is
how can i edit the dial plan so when i dial extension 1003 it calls
1003 right now i always get error code 483 to many hops?

On Apr 30, 2:30 pm, Fred Dixon <ffdi...@gmail.com> wrote:
> Hi Ryan,
>
> In BigBlueButton 0.8-beta-4, FreeSWITCH now only listens to the local
> loopback (127.0.0.1) address.  You can change this to listen to the
> external IP address (so FreeSWITCH can receive in coming calls).   See
>
>  http://code.google.com/p/bigbluebutton/issues/detail?id=1133
>
> Regards,... Fred
> --

> BigBlueButton Developerhttp://bigbluebutton.org/http://code.google.com/p/bigbluebutton

[Ryan Allen]

Well... I got it!!!! Not sure if there are any bugs yet or for a safe
stand point. Or if this is everything I tried a lot of stuff. But for
the newbie like me. If you want to use free switch to dial local sip
extensions here is how. Add the Hash Mod to your auto_load Configs.
and to your modxml file. Then add the following lines of code to your
default dial plan


<!--
dial the extension (1000-1019) for 30 seconds and go to voicemail if
the
call fails (continue_on_fail=true), otherwise hang up after a
successful
bridge (hangup_after_bridge=true)
-->
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])
$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app>
-->
<action application="bind_meta_app" data="1 b s execute_extension::dx
XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$$
{recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-
%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf
XML features"/>
<action application="bind_meta_app" data="4 b s
execute_extension::att_xfer XML features"/>
<action application="set" data="ringback=${us-ring}"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=30"/>
<!-- <action application="set" data="sip_exclude_contact=$
{network_addr}"/> -->
<action application="set" data="hangup_after_bridge=true"/>
<!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/
> -->
<action application="set" data="continue_on_fail=true"/>
<action application="hash" data="insert/${domain_name}-call_return/$
{dialed_extension}/${caller_id_number}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/$
{dialed_extension}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/$
{called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/
global/${uuid}"/>
<action application="set" data="called_party_callgroup=${user_data($
{dialed_extension}@${domain_name} var callgroup)}"/>
<!--<action application="export" data="nolocal:sip_secure_media=$
{user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/
>-->
<action application="hash" data="insert/${domain_name}-last_dial/$
{called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@$
{domain_name}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="loopback/app=voicemail:default $
{domain_name} ${dialed_extension}"/>
</condition>
</extension>
<!-- END freeswitch Stuff -->

<!-- BBB Stuff -->

<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>
<!-- END BBB Stuff -->



Now you need your sip phones to connect to your freeswitch internally.
So have your phones connect to port 5090. So use <serverip>:5090. That
should be it. refresh your xmls and clean bbb and should start up.
Then dial the default extensions or your 5 number bbb ID