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
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
