Hi, This sounds udp RTP problem. Might be you have some firewall rules that block this kind of traffic ? As soon I remember, Asterisk by default use random port between 10000 and 20000 for rtp traffic (you can adjust this in rtp.conf).
- Sebastien Jonathan Moore escribió: > 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 > _______________________________________________ -- 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
