[asterisk-dev] Asterisk 1.4-beta3 builtin transfer

[Volkov Alexei]

Hi!

As we discussed earlier, in asterisk 1.4-beta3 we need to define 
TRANSFER_CONTEXT just before Dial application to make builtin transfer 
work properly.

But lets assume following situation.
Caller A over h323 channel calling asterisk and routes to "B"  peer 
(SIP), while "B" is ringing he can determine "A" by his caller id and 
answers with this in mind. Then "B" wants transfer "A "to another SIP 
peer "C"  with attended transfer feature shortcut. "A" placed on hold 
and "C" starts ringing while "B" hearing ringing tones. In pre 1.4-beta3 
"C" peer supplied with caller id of "A", while 1.4-beta3 supplied with 
"s at TRANSFER_CONTEXT" caller id witch is wrong for me and i wants to see 
caller id of origin "A".

Is there some new configuration options in 1.4-beta3 making such a new 
behavior or it is an error?
How to make the same functionality was before 1.4-beta3?

WBR, Alexei Volkov.