Make sure you put t (for the called party) or T (for the calling party) in the Dial command option, like this Dial(somephone,timeout,tT)
On 4/11/05, Gonchi Mateos <[EMAIL PROTECTED]> wrote: > Hi all, > > We were willing to try the SIP Attended/Supervised transfer with * realease > 1.0-7. From the wiki�s feature.conf config page we found that a special > section called featuremap had to be added to the config: > > [featuremap] > blindxfer => #1 ; Blind transfer > disconnect => *0 ; Disconnect > automon => *1 ; One Touch Record > atxfer => *2 ; Attended transfer > > We made that changes but upon pressing *2 nothing happens, neither with > #1 for the blind transfer. The blind transfer is working as it defaults > in *, with # plus extension. > > We tried to unload and reload res_features module but with no luck as it > says that the user count is 1. > > After some examination at chan_sip.c, we found the supervised transfer code > section, but we found nothing on the parsing of the featuremap section. > We did find the parsing of the first section of the config file concerning > call parking, which does work. > > Any idea on how to make it work? > > Thanks to all, > Gonchi > > _______________________________________________ > 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
