Hi everyone. We have an asterisk server in our main office and phones at each remote site. The remote offices are connected via a MPLS which, to my knowledge has no natting going on.
The problem I have is that any call from a remote phone to a remote phone (even on the same remote lan) results in no audio. If I make a call from the same LAN the asterisk server is on, to one of these remote sites, I get perfect two way audio. If I play a call from one phone to another at a remote site, there is no audio, however, I do hear messages (such as voicemail, things from Playback(), etc) that originate on the asterisk server. I've tried adjusting canreinvite= in sip.conf in hopes in might have some effect, but so far nothing. Suggestions on where else to look, or what the problem might be? Which configs would be useful in troubleshooting? Thanks. -jonathan _______________________________________________ -- 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
