Don't hesitate anymore, use asterisk to do so with the
SetCallerID command in extensions.conf. You should
forward the call through Asterisk from SER. If you
don't want to have all the media pass through you, and
i beleive you don't, use canreinvite=yes in sip.conf,
you should fight a little bit with the codecs with
allow and disallow but it should be fine. The next
links could help you with the reinvitation issue:
http://www.voip-info.org/wiki-Asterisk+SIP+media+path
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
Hope it helps.
Andres
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