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From "Naderi, Sascha" <SNad...@datus.com>
Subject AW: SIP connectivity
Date Thu, 14 Feb 2013 07:09:50 GMT
Dear Maxim,





OK, thanks a lot. I will check it out and leave feedback.





Regards

Sascha

________________________________

Von: Maxim Solodovnik [solomax666@gmail.com]
Gesendet: Mittwoch, 13. Februar 2013 23:58
Bis: Naderi, Sascha
Cc: user@openmeetings.apache.org
Betreff: Re: SIP connectivity

please try red5sip rev. 76
it has additional parameter: om.context


On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha <SNaderi@datus.com<mailto:SNaderi@datus.com>>
wrote:

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<mailto: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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--
WBR
Maxim aka solomax




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WBR
Maxim aka solomax






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WBR
Maxim aka solomax




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WBR
Maxim aka solomax




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Maxim aka solomax

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