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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