I haven't changed rtp.conf from original installation. So the values are: rtpstart=10000 rtpend=20000
I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter "bindaddr" in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin <[EMAIL PROTECTED]> wrote:
what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 20000 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan <[EMAIL PROTECTED]> wrote: > Hi Koen > > > This works fine when I call this account from my personal gtalk. But others > > have some very strange problems. > > In most cases, I see the call coming into Asterisk and executing normally. > > On the callers side, the call looks like it was answered, but there's no > > audio. > > In some other cases, the call doesn't even appear to be answered, although I > > see a normal execution on Asterisk. > > Can you please open a bug report that describes your problem, and > attach an Asterisk debug output for a failed call to the report? > > Thanks, > > Philippe > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users