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From Bakko <asannu...@gmail.com>
Subject Re: SIP connectivity
Date Thu, 07 Feb 2013 15:07:34 GMT
Other think,

On the openmeetings table sipuser, field username is deprecated. Change 
to defaultuser.

Regards

El 07/02/2013 09:59, Maxim Solodovnik escribió:
> Just have tested it and it it works!
> Thanks a lot!
> I'll update the documentation
>
>
> On Fri, Feb 1, 2013 at 7:35 PM, Bakko <asannucci@gmail.com 
> <mailto:asannucci@gmail.com>> wrote:
>
>     Hello,
>
>     on Asterisk 1.8, extconfig.conf this line:
>
>     /sipusers => odbc,asterisk,sipusers
>
>     /is deprecated.
>
>     Now only use:
>
>     /sippeers => odbc,asterisk,sipusers
>
>     /Regards/
>
>     /
>     El 01/02/2013 00:00, Maxim Solodovnik escribió:
>>     I have updated the instruction (minor update)
>>
>>
>>     On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik
>>     <solomax666@gmail.com <mailto:solomax666@gmail.com>> 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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>>>>                                 message.
>>>>
>>>>
>>>>
>>>>                                 -- 
>>>>                                 WBR
>>>>                                 Maxim aka solomax
>>>>
>>>>
>>>>
>>>>                             -- 
>>>>                             WBR
>>>>                             Maxim aka solomax
>>>>
>>>
>>>
>>>
>>>
>>>                 -- 
>>>                 WBR
>>>                 Maxim aka solomax
>>
>>
>>
>>
>>         -- 
>>         WBR
>>         Maxim aka solomax
>>
>>
>>
>>
>>     -- 
>>     WBR
>>     Maxim aka solomax
>
>
>
>
> -- 
> WBR
> Maxim aka solomax


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