Openmeetings installed them for me, that's why I ended up with those.
Using the Asterisk ones makes more sense to me. Maybe it's a good idea
to have 'em removed from the install procedure.
BC
On 01/30/13 22:30, Jeff Clay wrote:
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
*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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