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 <dormiti...@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" <snad...@datus.com> > > An: "user@openmeetings.apache.org" <user@openmeetings.apache.org> > > CC: "solomax...@gmail.com" <solomax...@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 [solomax...@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 > > <snad...@datus.com<mailto:snad...@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 [solomax...@gmail.com<mailto:solomax...@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.conin...@telenet.be<mailto:bart.conin...@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 > > <solomax...@gmail.com<mailto:solomax...@gmail.com>> wrote: > > > > OK will add it and notify you > > > > On Jan 31, 2013 5:05 PM, "Bart Coninckx" > > <bart.conin...@telenet.be<mailto:bart.conin...@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.conin...@telenet.be<mailto:bart.conin...@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.conin...@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.conin...@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.conin...@telenet.be<mailto:bart.conin...@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.conin...@telenet.be<mailto:bart.conin...@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.conin...@telenet.be<mailto:bart.conin...@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 > 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 > > > > > > > > > > > > > > -- > > WBR > > Maxim aka solomax > > > > > > > > > > -- > > WBR > > Maxim aka solomax > > > > > > > > > > -- > > WBR > > Maxim aka solomax > -- WBR Maxim aka solomax
