Peter Fern wrote: > I've had the same problem with all boxen running the same version. We > ditched IAX2 for SIP and it has been working fine since. >
Well, upgrading my remote site to 1.2.5 appears to have fixed my issues. -Barry > Doug Lytle wrote: > >> Barry Flanagan wrote: >> >>> Hi, >>> >>> I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to >>> connect to a 1.2.5 box for PSTN. There are 15 users on the remote >>> server, all connecting via SIP softphones. >>> >>> For some reason, there is an increasing number of calls where the callee >>> does not get any audio although the caller can hear them perfectly. >>> >> >> I've had this problem in the past, when not running the same version >> of Asterisk on both ends of the trunk. >> >> Doug >> >> _______________________________________________ >> --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 -- -Barry Flanagan _______________________________________________ --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