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From Bart Coninckx <bart.conin...@telenet.be>
Subject Re: SIP connectivity
Date Wed, 30 Jan 2013 21:45:23 GMT
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.coninckx@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.coninckx@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.coninckx@telenet.be <mailto:bart.coninckx@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.coninckx@telenet.be
>             <mailto:bart.coninckx@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.coninckx@telenet.be
>             <mailto:bart.coninckx@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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>
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>
>
>
>             -- 
>             WBR
>             Maxim aka solomax
>
>
>
>         -- 
>         WBR
>         Maxim aka solomax
>


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