[asterisk-users] SIPAddHeader back to source
webaccounts at jgoettgens.de
Mon Mar 17 12:04:03 CDT 2014
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.
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