[Asterisk-Dev] sip refer

[Richard]

Hi,

When * gets a sip REFER message, it looks through the context and dial plan.
Since I use * as a PSTN gateway with ser, the desired effect would be *
sending a sip INVITE message to the "Refer-To" address inside the sip REFER
message.

In case of a blind transfer, can I just modify chan_sip.c to make a new sip
call and replace the existing call with it? Anything I need to worry about
this approach?

Thanks,

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