[Asterisk-Users] Is there a way to avoid bandwidth consumption on sip calls?

[Helder Rogerio [MICROREDE]]

Hi!

Is there a way to avoid being "at the middle" of communications between
two SIP endpoints? So that we can avoid loosing bandwidth with it?

Is there a way to "forward" the authentication to a IAX provider and
"transfer" the call to it, avoiding using my own bandwidth?


I've tested it with SER with some results, I was wondering if it is
possible with Asterisk.


Cumprimentos / Best regards,

Helder Rogério


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