Bart,
If you look in the source directory of your asterisk tar
file, under contrib/realtime/mysql you’ll find the .sql
files required for all the realtime drivers. I never
thought to use the ones with OM.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
*From:*Bart Coninckx [mailto:bart.conin...@telenet.be]
*Sent:* Wednesday, January 30, 2013 3:19 PM
*To:* user@openmeetings.apache.org
<mailto:user@openmeetings.apache.org>
*Cc:* Jeff Clay
*Subject:* Re: SIP connectivity
Well,
I might have found one difference though:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
dictates how the table should look like. I obviously
used the one in the openmeetings mysql database, but
this one seems to miss the table "useragent". I
discovered this because it showed up in the logfiles.
BC
On 01/29/13 14:41, Jeff Clay wrote:
Bart,
From an asterisk configuration standpoint there are
very few differences between 1.8.x and 11.x. If
memory serves, the only major changes that I ran
into (in my production environment) was changes to
SIP NAT values and the behavior of app_page() now
uses confbridge instead of meetme to mix the audio.
Also, TCP, TLS and app_confbridge got a major
overhauling. There were of course many other changes
and bug fixes, you can skim through the change log
for full details, but I think that was the jist of it.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
*From:*Bart Coninckx [mailto:bart.conin...@telenet.be]
*Sent:* Tuesday, January 29, 2013 4:02 AM
*To:* Maxim Solodovnik
*Cc:* user
*Subject:* Re: SIP connectivity
I see - I'm willing to try the 11 version in the
next fiew days if desired.
BC
On 01/29/13 10:57, Maxim Solodovnik wrote:
I test the integration using
Asterisk 1.8.13.1 (Ubuntu 12.10)
On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx
<bart.conin...@telenet.be
<mailto:bart.conin...@telenet.be>> wrote:
That is amazing - I initially tried to do the
same thing by using the new chan_motif driver in
Asterisk 11 which connects to a XMPP server.
Are you guys using Asterisk 11? This version is
the newest LTS version and has the best video
capabilities.
Cheers,
BC
On 01/29/13 02:44, Maxim Solodovnik wrote:
red5sip will create special OM user in the
room: "SIP Transport"
after that you can call to the OM room using
SIP hard or soft phone.
We are currently testing it and trying to
add video capabilities ...
On Tue, Jan 29, 2013 at 4:44 AM, Bart
Coninckx <bart.conin...@telenet.be
<mailto:bart.conin...@telenet.be>> wrote:
Hi Jeff,
In fact, I saw both pages, but none explain
what they set up to do, just a bunch of
command line instructions are given.
Your "OM will create a meetme meeting as
configured in the realtime meetme database"
actually says it all in one go :-)
cheers,
BC
On 01/28/13 22:38, Jeff Clay wrote:
Bart,
OM will create a meetme meeting as
configured in the realtime meetme database.
Have you read this page
https://cwiki.apache.org/OPENMEETINGS/openmeetings-asterisk-integration.html
? You might also check out
http://openmeetings.apache.org/red5sip-integration.html
but I assume this is the one you're already
referring to.
Jeff Clay
Network Administrator
Infotech Enterprises America
870-215-5506
Ext. 1506
-----Original Message-----
From: Bart Coninckx
[mailto:bart.conin...@telenet.be
<mailto:bart.conin...@telenet.be>]
Sent: Monday, January 28, 2013 3:36 PM
To: user@openmeetings.apache.org
<mailto:user@openmeetings.apache.org>
Subject: SIP connectivity
Hi,
I noticed some documentation on how to
connect OM with a SIP proxy or server, more
particularly with the MeetMe application in
Asterisk.
The exact goal or purpose is not mentionned
however. Will OM callout to a MeetMe
conference? Or is it the other way round?
Cheers,
Bc
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WBR
Maxim aka solomax
