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
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