[asterisk-users] Linksys 942 Call Transfer

[joakimsen at gmail.com]

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!)