I do not believe that the asterisk context is related to the url of openmeetings.
Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 From: Naderi, Sascha [mailto:snad...@datus.com] Sent: Wednesday, February 13, 2013 2:00 PM To: user@openmeetings.apache.org Cc: Maxim Solodovnik [solomax...@gmail.com] Subject: Re: SIP connectivity 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] 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
