My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, "meetme show <conf>" does show the SIP caller as "talking" when they do.
Here's the current channels when the conference is up in that configuration: asterisk*CLI> core show channels Channel Location State Application(Data) DAHDI/23-1 2...@conferences:2 Up MeetMe(201,cosT) DAHDI/pseudo-1338070 s...@default:1 Rsrvd (None) SIP/150-b444d988 2...@conferences:2 Up MeetMe(201,cosT) What should I be looking at to debug this? _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
