Just have tested it and it it works! Thanks a lot! I'll update the documentation
On Fri, Feb 1, 2013 at 7:35 PM, Bakko <asannu...@gmail.com> wrote: > Hello, > > on Asterisk 1.8, extconfig.conf this line: > > *sipusers => odbc,asterisk,sipusers > > *is deprecated. > > Now only use: > > *sippeers => odbc,asterisk,sipusers > > *Regards* > > * > El 01/02/2013 00:00, Maxim Solodovnik escribió: > > I have updated the instruction (minor update) > > > On Thu, Jan 31, 2013 at 7:05 PM, Maxim Solodovnik <solomax...@gmail.com>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 >> > > > > -- > WBR > Maxim aka solomax > > > -- WBR Maxim aka solomax
