asterisk users - Nov 2007 - Linksys 942 Call Transfer

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






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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!)