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
