This is my issue as well, Does anyone know how to fix it? Thanks, Michael
On Mon, 29 Sep 2003, Dave Weis wrote: > > On Mon, 29 Sep 2003, Brian Capouch wrote: > > Christopher J. Wolff wrote: > > > Is it safe to assume that a fresh CVS build will not have the SIP > > > translation problem described? > > > > > > Just FYI: I had similar problems for a while, and then I completely > > > scrapped my CVS directory and did a CVS CHECKOUT (instead of an update). > > > That solved the problem. > > > > I "checkout" rebuilt from CVS last night about 10:00 pm, and filed a bug > > report at that time. > > It isn't *all* SIP calls for me, btw; my ATA186 works just fine, but my > > Budgetone won't work with the "broken" code. . . > > It's as described in other mails--I can receive calls on the Budgetone > > but when I make them the RTP part is broken and the calls cut off the > > second they're set up. > > I'm seeing the same thing with my budgetones and today's cvs. They are all > on the same network and worked on previous versions. I can call > voicemailmain and the console says that it is playing but I hear no sound. > Then it hangs up automatically. > > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
