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From Bart Coninckx <bart.conin...@telenet.be>
Subject Re: SIP connectivity
Date Thu, 31 Jan 2013 10:05:16 GMT
It is for Asterisk 11 - don't know for other versions. You probably have 
no issues because of the 1.8 version. To be sure the .sql files in the 
Asterisk source should be compared across versions.

this one is missing:

`useragent` varchar(20) DEFAULT NULL,

complete list (I think)  is on:

https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure


If I bump into others, I'll report ASAP,


BC



On 01/31/13 06:21, Maxim Solodovnik wrote:
> Is the OM meetme table incomplete?
> My asterisk reports no issues :(
>
> could you provide me with missing fields and I'll add it.
> My purpose was to create table with required fields only.
>
>
> On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx 
> <bart.coninckx@telenet.be <mailto:bart.coninckx@telenet.be>> wrote:
>
>     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
>>     <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.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
>>
>
>
>
>
> -- 
> WBR
> Maxim aka solomax


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