Hi
I have a customer who is using Linksys 942 phones.
When they try to transfer a call the Asterisk CLI
reports that both legs of the call must exist on the
server. The call they are trying to transfer then
drops.
Does anyone know why this is and how to fix it?
TIA
Regards
Jon
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
