Invite with replaces limitation for sip voice gateways

Unanswered Question
Oct 16th, 2009
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Reading this paper from Cisco's web site,


http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps6831/product_data_sheet0900aecd804110a2.html


I found in the Standards Support part, regarding rfc 3892 this foot note:

"7 Invite with Replace implementation is limited to transfer."


Does anybody know if extending this feature to other call flows is in the road map of any development path? Particularly I see this feature useful to enable allowing call pickup of calls from pstn to sip,for example, which requires accepting an Invite with replaces header without previous refer transaction involved.

Thanks

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