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
