I'm using SIP INFO and ulaw. It seems that the same thing happens from SIP to SIP as well, regardless of what the DTMF setting is. The actual problem is that the calling user can transfer, but the called user cannot. I just tried the latest CVS snapshot and the v1.0 stable branch and they both behave the same way.
[EMAIL PROTECTED] wrote: > Maybe you are using inband DTMF with a compressed codec. DTMF on a > call with any codec other than ulaw or alaw MUST use OOB DTMF like > RFC2833 or INFO. > > On Fri, 2004-03-05 at 20:39, Barton Hodges wrote: >> I'm having a problem with transferring a call that comes in a Zap >> channel and is connected with a SIP channel (on a GS HT-286). >> >> The call is answered automatically, then the user enters an >> extension. Dial() is called with both T and t flags. When the >> bridge is made between the channels, the caller on the Zap channel >> can hit '#' to transfer, but the caller on the SIP channel cannot. >> No messages whatsoever are displayed on the console when the SIP >> user hits any keys. What am I missing? >> >> >> _______________________________________________ >> 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 _______________________________________________ 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
