Set canreinvite=no in sip.conf to force the RTP voice traffic to pass through asterisk so it can do the transcoding.
-Seth On Tue, 2004-07-13 at 10:46, James Dutton wrote: > Hi > > Further to my previous email... > > I have a Xten software phone connecting to a Grandstream 100 hardware > phone. My first problem is that voice transmits in one direction only. > Secondly, this only works if the codecs on both are identical. If the > Xten uses GSM and the Grandstream uses ULAW then the phones connect, > but no voice can be hears in either direction. I assumed (possibly > wrongly) that Asterisk did the appropriate codec translation? > > Regards > James -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users