Return-Path: X-Original-To: apmail-openmeetings-user-archive@www.apache.org Delivered-To: apmail-openmeetings-user-archive@www.apache.org Received: from mail.apache.org (hermes.apache.org [140.211.11.3]) by minotaur.apache.org (Postfix) with SMTP id B0F30EA38 for ; Wed, 30 Jan 2013 21:45:52 +0000 (UTC) Received: (qmail 97505 invoked by uid 500); 30 Jan 2013 21:45:52 -0000 Delivered-To: apmail-openmeetings-user-archive@openmeetings.apache.org Received: (qmail 97481 invoked by uid 500); 30 Jan 2013 21:45:52 -0000 Mailing-List: contact user-help@openmeetings.apache.org; run by ezmlm Precedence: bulk List-Help: List-Unsubscribe: List-Post: List-Id: Reply-To: user@openmeetings.apache.org Delivered-To: mailing list user@openmeetings.apache.org Received: (qmail 97466 invoked by uid 99); 30 Jan 2013 21:45:52 -0000 Received: from athena.apache.org (HELO athena.apache.org) (140.211.11.136) by apache.org (qpsmtpd/0.29) with ESMTP; Wed, 30 Jan 2013 21:45:52 +0000 X-ASF-Spam-Status: No, hits=2.2 required=5.0 tests=HTML_MESSAGE,RCVD_IN_DNSWL_NONE,SPF_PASS X-Spam-Check-By: apache.org Received-SPF: pass (athena.apache.org: local policy) Received: from [195.130.137.68] (HELO georges.telenet-ops.be) (195.130.137.68) by apache.org (qpsmtpd/0.29) with ESMTP; Wed, 30 Jan 2013 21:45:46 +0000 Received: from [192.168.1.20] ([84.196.1.254]) by georges.telenet-ops.be with bizsmtp id uMlP1k00E5UpHxg06MlPyU; Wed, 30 Jan 2013 22:45:23 +0100 Message-ID: <51099473.9010101@telenet.be> Date: Wed, 30 Jan 2013 22:45:23 +0100 From: Bart Coninckx User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:17.0) Gecko/20130105 Thunderbird/17.0.2 MIME-Version: 1.0 To: user@openmeetings.apache.org CC: Jeff Clay Subject: Re: SIP connectivity References: <5106EF5D.5030601@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCBAFA@INFHYDEX10MB2.corp.infotech-enterprises.com> <5106F125.4020502@telenet.be> <51079B9A.8050108@telenet.be> <51079E2D.1080004@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCBDE2@INFHYDEX10MB2.corp.infotech-enterprises.com> <51098E31.7010604@telenet.be> <2B93465B6F724E45BCF490E30DA2FF6701FCCB04@INFHYDEX10MB2.corp.infotech-enterprises.com> In-Reply-To: <2B93465B6F724E45BCF490E30DA2FF6701FCCB04@INFHYDEX10MB2.corp.infotech-enterprises.com> Content-Type: multipart/alternative; boundary="------------040600040004030903080600" X-Virus-Checked: Checked by ClamAV on apache.org This is a multi-part message in MIME format. --------------040600040004030903080600 Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit 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 > *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 > > > 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 > > 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 > > ________________________________ > > DISCLAIMER: > > This email may contain confidential information and is > intended only for the use of the specific individual(s) to > which it is addressed. If you are not the intended > recipient of this email, you are hereby notified that any > unauthorized use, dissemination or copying of this email > or the information contained in it or attached to it is > strictly prohibited. If you 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 > --------------040600040004030903080600 Content-Type: text/html; charset=UTF-8 Content-Transfer-Encoding: 8bit
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
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> 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

________________________________

DISCLAIMER:

This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you 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

 

 


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