did you try canreinvite=no
in your sip.conf file It would also help to: 1) Post the relevant configuration files (phone AND Asterisk) 2) Post the EXACT message from column 1 to EOL 3) What version of Asterisk? Stock? From a certain distribution? Patches? Or I could just say "There is a problem with your configuration, transfer of calls from an SPA-phone works fine for me." (it really does!) _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
