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From Bakko <asannu...@gmail.com>
Subject Re: SIP connectivity
Date Wed, 13 Feb 2013 22:04:44 GMT
Hello,

two questions.

Asterisk integration working on openmeetings 2.0?

Where can I download 2.1 version

Thank you

Regards


El 13/02/2013 14:59, Naderi, Sascha escribió:
>
> Dear all,
>
> i have tested the asterisk sip integration as documented with the most recent instruction

> (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works 
> just fine.
>
> The only thing i am missing is a way to get this working when 
> i choose to rename the openmeetings context from 
> http://yourcorp.com:5080/openmeetings  to 
> http://yourcorp.com:5080/yourmeetings
>
> Which settings do i have to modify 
> so that red5sip functions even if the context name is changed?
>
> Regards
> Sascha Naderi
>
>
> ------------------------------------------------------------------------
> *Von:* Maxim Solodovnik [solomax666@gmail.com]
> *Gesendet:* Samstag, 9. Februar 2013 02:32
> *Bis:* Bart Coninckx
> *Cc:* user
> *Betreff:* Re: SIP connectivity
>
> All tables are created by OM automatically
>
> On Feb 9, 2013 5:46 AM, "Bart Coninckx" <bart.coninckx@telenet.be 
> <mailto: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 <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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>>>>                             the use of the specific individual(s)
>>>>                             to which it is addressed. If you are
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>>>>                             email, you are hereby notified that any
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>>>>                             received this message in error, please
>>>>                             immediately notify the sender at
>>>>                             Infotech and delete the original message.
>>>>
>>>>
>>>>
>>>>                             -- 
>>>>                             WBR
>>>>                             Maxim aka solomax
>>>>
>>>>
>>>>
>>>>                         -- 
>>>>                         WBR
>>>>                         Maxim aka solomax
>>>>
>>>
>>>
>>>
>>>
>>>             -- 
>>>             WBR
>>>             Maxim aka solomax
>>
>>
>>
>>
>>     -- 
>>     WBR
>>     Maxim aka solomax
>


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