MURALI V
2010-Mar-10 06:31 UTC
[asterisk-users] call features affected by native bridging between sip phones
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......... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100310/c7a0fa20/attachment.htm