Tony Graziano wrote: [...] > I restarted freeswitch service from sipxconfig. > > I connected to the test unit from outside with xlite via sipxbridge. I > have two systems, a test unit (3.11.3, dev with freeswitch), and a > limited production unit (3.10.1) that can dial each other natively. The > production unit uses an Ingate firewall. > >>From the outside I can call from a polycom handset through the Ingate on > the production system to the conference on the dev system. From the PC > next to it (from the outside), I register and dial to the conference > through the dev system. Wow, that was cool. No audio problems but will > do more testing next week. Can hear each user enter the conference, and > leave and know if I am only one in the conference. > > Since I was on "both" phones, I was the only one there, and was talking > to myself. Once I started feeling stupid (didn't take long) I hung up > (saw CDR's generate fine) and decided it's saner to test with more than > just me. I talk to myself too much already. >
Nice! I spent a few minutes talking to myself on two phones as well :) Now to get it fixed so it works from sipXconfig. If I get any time this weekend I'll take a look at it, but I think I'll be trying to finish up 3.11.3 issues too... ah, so much to do. I also spent some time playing with the new conference management tools that Joe implemented. I don't think these have made it to a nightly build yet due to a problem building RPMs that I need to take a look at. Very nice work by Joe though.. one of the nicest pages in sipXconfig. Kevin _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
