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From Bakko <asannu...@gmail.com>
Subject Re: SIP connectivity
Date Fri, 01 Feb 2013 12:35:04 GMT
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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>>>
>>>
>>>
>>>                             -- 
>>>                             WBR
>>>                             Maxim aka solomax
>>>
>>>
>>>
>>>                         -- 
>>>                         WBR
>>>                         Maxim aka solomax
>>>
>>
>>
>>
>>
>>             -- 
>>             WBR
>>             Maxim aka solomax
>
>
>
>
>     -- 
>     WBR
>     Maxim aka solomax
>
>
>
>
> -- 
> WBR
> Maxim aka solomax


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