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From Maxim Solodovnik <solomax...@gmail.com>
Subject Re: SIP connectivity
Date Tue, 19 Feb 2013 13:44:33 GMT
You will need to set up red5sip separately (it licence is incompatible with
Apache)


On Tue, Feb 19, 2013 at 8:42 PM, BBS Technik <dormitilla@gmx.de> wrote:

> Hello,
>
> that are very good news. Could one of the sip experts document the finaly
> needed steps in the VOIP / SIP Integration documentation in the wiki?
>
> Will the upcomming 2.1 release support the sip integration out of the box?
>
> Best regards
> Ed
>
>
>
> -------- Original-Nachricht --------
> > Datum: Tue, 19 Feb 2013 13:01:03 +0000
> > Von: "Naderi, Sascha" <SNaderi@datus.com>
> > An: "user@openmeetings.apache.org" <user@openmeetings.apache.org>
> > CC: "solomax666@gmail.com" <solomax666@gmail.com>
> > Betreff: Re: SIP connectivity
>
> > Dear Maxim, dear all,
> >
> >
> >
> >
> >
> > i tried it with the latest red5sip rev. (91) and it worked fine with a
> > changed openmeetings context.
> >
> > Thank you!
> >
> >
> >
> >
> >
> >
> >
> > Regards
> >
> > Sascha
> >
> > ________________________________
> >
> > Von: Naderi, Sascha
> > Gesendet: Donnerstag, 14. Februar 2013 08:09
> > Bis: Maxim Solodovnik
> > Cc: user@openmeetings.apache.org
> > Betreff: AW: SIP connectivity
> >
> >
> > 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
> >
> > ________________________________
> >
> > 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
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> > 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
> >
> >
> >
> >
> >
> >
> > --
> > WBR
> > Maxim aka solomax
> >
> >
> >
> >
> > --
> > WBR
> > Maxim aka solomax
> >
> >
> >
> >
> > --
> > WBR
> > Maxim aka solomax
>



-- 
WBR
Maxim aka solomax

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