[asterisk-users] SIPAddHeader back to source

[jg]

Hi,

I am using the XML-browser and Call-Info header features for some SIP phones. 
SIPAddHeader(Call-Info: ...) seems to work only in the outgoing direction. Does somebody know a 
way to send a Call-Info header to the originating SIP device by using only the dial plan? 
Currently, I am using the XML-browsers to update callee info, but I'd like to use the icon 
purpose to do that.

It's probably easier to embed this functionality into a CTI application using an AMI command 
like Originate (such that internally Dial() gets called twice), but this triggers it from the 
outside. sipsak could be called from extensions.conf, but I'd like to avoid that. Transferring 
to a Local channel after entering the dial plan might also work, but that looks clumsy. I am 
sorry if I have overlooked a standard method to send a header back to the source.

jg