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:[email protected]]
*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
<[email protected] <mailto:[email protected]>> 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
<[email protected] <mailto:[email protected]>>
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:[email protected]
<mailto:[email protected]>]
Sent: Monday, January 28, 2013 3:36 PM
To: [email protected]
<mailto:[email protected]>
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
________________________________
DISCLAIMER:
This email may contain confidential information and is
intended only for the use of the specific individual(s) to
which it is addressed. If you are not the intended recipient
of this email, you are hereby notified that any unauthorized
use, dissemination or copying of this email or the information
contained in it or attached to it is strictly prohibited. If
you received this message in error, please immediately notify
the sender at Infotech and delete the original message.
--
WBR
Maxim aka solomax
--
WBR
Maxim aka solomax
