Trunking (IAX/SIP) Between external Asterisk and BBB
sw4pon
i have external asterisk (1.4) server, i want to setup a trunk between
the existing asterisk PBX and the BBB box in order to get a extension
number to dial from the internal dialplan to get in the conference
(MeetMe) hosted into BBB.
I was able to setup different trunks for testing (iax2/sip) between
the
systems and the needed outbound routes, but the real question/problem
still is:
How to configure the BBB's asterisk in order to get an extension that
routes (the routed via trunk number) to the meeting?
I think that i need on BBB:
- a local extension (say: 748 for example).
- "something" to reroute the given extension into the meeting.
btw thanks for this amazing piece of software!
Thanks
nathana
pretty much wanting to do the same thing. Can you share with me how
you got the two asterisks linked? I haven't linked the two yet here
and haven't ever had a reason to do it yet. Anything you could offer
would be much appreciated. thanks!
Richard Alam
Just edit /usr/share/red5/webapps/bigbluebutton/WEB-INF/bigbluebutton.properties
and usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties to
point to your asterisk server instead of the one that came with BBBB.
Richard
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BigBlueButton
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nathana
So what you are saying, just to clarify. Is that We can point BBB
voice module to use an existing asterisk server that uses meete or
konference by modifying those files. Then BBB will just use that
server instead of the included asterisk build. And one can map our
existing dial plans to the bbb meeting room(s).
From what you posted this is my understanding on what needs to be
done? If I am missing anything please let me know.
1. Modify /usr/share/red5/webapps/bigbluebutton/WEB-INF/
bigbluebutton.properties
a. Run create-sip-users.sh to create file bbb_sip.conf
This file creates 100 asterisk extensions that bbb uses for meetme
conference
2. Modify usr/share/red5/webapps/sip/WEB-INF/ bigbluebutton-
sip.properties
a. Edit Asterisk Management Interface login values
b. change conference application from konference to meetme
3. modify external asterisk server sip.conf adding bbb_sip.conf
; comment: this is how sip.conf might look on external asterisk
server
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include bbb_sip.conf
4. Add bbb-voice-conference\config\asterisk\bbb_extensions.conf to
external asterisk server extensions.conf file.
Questions I have are:
How does BBB determine what conference to join? Does BBB interact with
the meetme database? How does from a 50,000 foot level view if there
are more than 1 rooms going?
Thanks!
Richard Alam
0.7 you only need one user (bbbuser).
Make sure that app_konference you use is from
http://github.com/jthomerson/AsteriskAudioKonf as it has modifications
to send messages that BBB understands.
BBB doesn't do anything with meetme database. It's up to your dialplan
to handle calls coming from BBB in the context you defined for
bbbuser. The "meetme" or "konference" entry in
bigbluebutton.properties is just a hint to BBB so it can parse
messages it receives from Asterisk AMI.
nathana
I was looking at the source download on git. So some of the files and
comments in them might not be accurate. thats ok.
I'll try it out. Thanks for the feedback.
On Oct 13, 1:58 pm, Richard Alam <ritza...@gmail.com> wrote:
> From your description, you seem to be using 0.64 version of BBB. In
> 0.7 you only need one user (bbbuser).
>
Richard Alam
know so we can fix.
Thanks.
nathana
http://github.com/bigbluebutton/bigbluebutton/blob/master/bbb-voice-conference/config/asterisk/create-sip-users.sh#
is still referenced there and in the files.
In bluebutton-sip.properties its in the comments.
# The start/stop RTP port the application is going to use
# for the media stream.
# NOTE: This will also be used as SIP users to REGISTER with
# Asterisk. Therefore, make sure you have this range of users
# in your bbb_sip.conf.
# See http://code.google.com/p/bigbluebutton/source/browse/#svn/trunk/bbb-voice-conference/config/asterisk
# create-sip-users.sh script to create the users.
startAudioPort=34380
stopAudioPort=34479
So if its not an IDOT error on my part, which is possibel you might
want to have someone clean that up for .7 and beyond. If I'm an idot
please let me know too. lol
nathana
in an incorrect ip address for my external asterisk server, the the
user interface bombed, saying couldn't connect. So I was making
progress. When I tried using the correct IP of the external asterisk
server, ui works. I could join the audio conference.
Looking at the asterisk logging I wasn't seeing any activity. So I
stopped the asterisk service on BBB. Then audio was dead. So even
though it was pointing and connecting using the modified files
bigbluebutton-sip.properties and bigbluebutton.properties. It is still
using the asterisk build in BBB local.
Any ideas on what else needs to be modified in BBB. Seems like I got
1/2 of it. Theoretically I would think if it is set right I could kill
bbb's included asterisk service?
Thanks!
Richard Alam
On Wed, Oct 13, 2010 at 10:57 PM, nathana <nat...@powertronglobal.com> wrote:
> So I tried this out modifying the files you suggested. At first I put
> in an incorrect ip address for my external asterisk server, the the
> user interface bombed, saying couldn't connect. So I was making
> progress. When I tried using the correct IP of the external asterisk
> server, ui works. I could join the audio conference.
>
> Looking at the asterisk logging I wasn't seeing any activity. So I
> stopped the asterisk service on BBB. Then audio was dead. So even
> though it was pointing and connecting using the modified files
> bigbluebutton-sip.properties and bigbluebutton.properties. It is still
> using the asterisk build in BBB local.
>
Did you restart Red5 (sudo /etc/init.d/red5 restart) after you
modified the files? You should for the properties to take effect.
Also, take a look at /usr/share/red5/log/sip.log and
/usr/share/red5/log/event*log for clues.
> Any ideas on what else needs to be modified in BBB. Seems like I got
> 1/2 of it. Theoretically I would think if it is set right I could kill
> bbb's included asterisk service?
>
Yes you can permanently stop the Asterisk that came with BBB and use
your own Asterisk.
> Thanks!
nathana
2010-10-14 17:30:13,109 [Thread-5] ERROR o.b.voiceconf.sip.SipPeer -
Failed to register with Sip Server in Sip.log
I noticed the default sip port in BBB was 5070. My asterisk server is
set to 5060. So I set sip.server.port=5060 in bigbluebutton-
sip.properties.
If I do a sip degug on in asterisk, I never see it try to connect.
Would I need to change this port rage in BBB somewhere? Is there some
internal firewall in BBB or port range that needs to be set elsewhere?
Thanks for the help.
Richard Alam
> This is the error I'm getting.
>
> 2010-10-14 17:30:13,109 [Thread-5] ERROR o.b.voiceconf.sip.SipPeer -
> Failed to register with Sip Server in Sip.log
>
BBB failed to register with Asterisk. Make use the username and
password matches in /etc/asterisk/bbb_sip.conf and
/usr/shared/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties
> I noticed the default sip port in BBB was 5070. My asterisk server is
> set to 5060. So I set sip.server.port=5060 in bigbluebutton-
> sip.properties.
>
port 5070 is ok...basically this is the SIP port for BBB SIP component
while Asterisk uses 5060.
> If I do a sip degug on in asterisk, I never see it try to connect.
> Would I need to change this port rage in BBB somewhere? Is there some
> internal firewall in BBB or port range that needs to be set elsewhere?
>
> Thanks for the help.
>
>
>
sw4pon
but actually i prefer a sip/iax2 trunk between these system because i
*dont want* the bbb box
to reach the PBX directly (different zones, different filtering
policies, and so on).
Btw yesterday i was able to get a working (but maybe dirty) setup,
my approach was:
* Create a sip/iax2 trunk between the PBX and BBB (refer to the
asterisk docs).
* Add an "Outbound Route" on your PBX for a "non existant" internal
number in your current
dialplans ( say 800), specify the previously created trunk as
preferred trunk for this route).
* On BBB edit "/etc/asterisk/bbb_extensions.conf" and add an extension
for your "non existant"
internal number, do what you want with it, but at least specify
something to reach an active
MeetMe a dirty example should be something like:
" exten => 800,1,Goto(prompt,s,1) ".
Now, if someone calls the internal number 800 the call should be
forwarded from your PBX to
the BBB's asterisk (thanks to the Trunk) the call now should be routed
to an existing MeetMe
Conference (thanks to the exten).
Please note that this is only a starting point, for informations about
the possible MeetME
"discovery strategy" have a look to the existing extensions in "/etc/
asterisk/bbb_extensions.conf".
Regards
(xdr)
> You don't need to peer the two asterisk.
>
> Just edit /usr/share/red5/webapps/bigbluebutton/WEB-INF/bigbluebutton.properties
> and usr/share/red5/webapps/sip/WEB-INF/bigbluebutton-sip.properties to
> point to your asterisk server instead of the one that came with BBBB.
>
> Richard
>
>
>
>
>
> On Wed, Oct 13, 2010 at 11:48 AM, nathana <nath...@powertronglobal.com> wrote:
> > I also have 1.4 . You are a little father ahead of me with BBB. I'm
> > pretty much wanting to do the same thing. Can you share with me how
> > you got the two asterisks linked? I haven't linked the two yet here
> > and haven't ever had a reason to do it yet. Anything you could offer
> > would be much appreciated. thanks!
>
> >>>I was able to setup different trunks for testing (iax2/sip) between the systems
>
> > --
> > You received this message because you are subscribed to the Google Groups "BigBlueButton-Setup" group.
> > To post to this group, send email to bigbluebu...@googlegroups.com.
> > To unsubscribe from this group, send email to bigbluebutton-s...@googlegroups.com.