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From Maxim Solodovnik <solomax...@gmail.com>
Subject Re: SIP connectivity
Date Thu, 07 Feb 2013 15:12:37 GMT
We already set two columns: name and defaultuser:
https://svn.apache.org/repos/asf/openmeetings/trunk/singlewebapp/src/org/apache/openmeetings/persistence/beans/sip/asterisk/AsteriskSipUser.java


On Thu, Feb 7, 2013 at 10:07 PM, Bakko <asannucci@gmail.com> wrote:

>  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> 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>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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>>>>>>
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If you
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>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> WBR
>>>>>> Maxim aka solomax
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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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