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From "BBS Technik" <dormiti...@gmx.de>
Subject Re: SIP connectivity
Date Tue, 19 Feb 2013 13:42:03 GMT
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
> 
> ________________________________
> 
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> 
> 
> 
> 
> --
> WBR
> Maxim aka solomax
> 
> 
> 
> 
> --
> WBR
> Maxim aka solomax
> 
> 
> 
> 
> 
> 
> --
> WBR
> Maxim aka solomax
> 
> 
> 
> 
> --
> WBR
> Maxim aka solomax
> 
> 
> 
> 
> --
> WBR
> Maxim aka solomax

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