Hi all,
I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established.
Is there a way that call setup is established, the RTP stream pass just between the SIP endpoints.
Example:
Works like this
SIP IP phones <-----------Asterisk RTP stream--------------> SIP IP phone
Asterisk
SIP IP phones <------------------RTP------------------------> SIP IP phone
Thanks!
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