OK will add it and notify you On Jan 31, 2013 5:05 PM, "Bart Coninckx" <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>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 > > >
