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From Bart Coninckx <bart.conin...@telenet.be>
Subject Re: SIP connectivity
Date Fri, 08 Feb 2013 22:46:53 GMT
May I add that a portion is missing, since one explains how to configure 
Asterisk for Realtime, but one does not stipulate how to create the 
necessary tables.
It's in my CentOS docs however (which I hope to post shortly).

BC

On 01/31/13 13:05, Maxim Solodovnik wrote:
> Hello Bart,
>
> I just take a look at your URL ...
> OM does not create/use sipfriends DB table (at least from version 2.1)
> only meetme table is used
>
> so I'm afraid there is nothing to change here
>
> Here is the most recent instruction:
> http://openmeetings.apache.org/red5sip-integration_2.1.html
>
> Will ask our SIP guru to review it one more time :)
>
>
>
> On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik 
> <solomax666@gmail.com <mailto:solomax666@gmail.com>> wrote:
>
>     OK will add it and notify you
>
>     On Jan 31, 2013 5:05 PM, "Bart Coninckx" <bart.coninckx@telenet.be
>     <mailto:bart.coninckx@telenet.be>> wrote:
>
>         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
>
>
>
>
> -- 
> WBR
> Maxim aka solomax


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