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From Maxim Solodovnik <solomax...@gmail.com>
Subject Re: SIP connectivity
Date Sat, 09 Feb 2013 01:32:29 GMT
All tables are created by OM automatically
On Feb 9, 2013 5:46 AM, "Bart Coninckx" <bart.coninckx@telenet.be> wrote:

>  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>wrote:
>
>> OK will add it and notify you
>>  On Jan 31, 2013 5:05 PM, "Bart Coninckx" <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
>>> > 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<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<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> 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> 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]
>>>> Sent: Monday, January 28, 2013 3:36 PM
>>>> To: 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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>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> WBR
>>>> Maxim aka solomax
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>>  --
>>> WBR
>>> Maxim aka solomax
>>>
>>>
>>>
>
>
>  --
> WBR
> Maxim aka solomax
>
>
>

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