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>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<bart.conin...@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.conin...@telenet.be<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> > 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> > 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] > 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 **** > > **** > > ** ** > > > -- WBR Maxim aka solomax
