[asterisk-users] call features affected by native bridging between sip phones

[MURALI V]

Hi Geeks,

       I am a beginner in asterisk, I read about native bridging option in
asterisk which allows the RTP streaming through the SIP media terminals
after initiating the call . I identified the following features are getting
affected
by this feature in my testing.

 1) Call transfer.
 2) Music On Hold
 3) Conferencing with meetme.

    I wonder if there are any other features will get affected due to native
bridging. Thanks in advance.

Regards

Murali Vasu

-- 
Smile is the only priceless gift you can give without a price.........
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