[asterisk-dev] CallerID retain on internal transfer

[Olivier Krief]

Hi,

>From http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/148565/focus=149455, you can read that :
- SIP allows CallerID to be changed at the point when 2 separate calls are bridged to one ... 
- May 2006 trunk version of Asterisk did not support this behaviour at that time.

Is it still true today ?
Is this feature considered for inclusion in 1.4 or 1.6 development cycle ?

Regards
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