All tables are created by OM automatically On Feb 9, 2013 5:46 AM, "Bart Coninckx" <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>wrote: > >> 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 >>> >>> >>> > > > -- > WBR > Maxim aka solomax > > >
