Some operators configure their SIP servers so that the RTP traffic does not pass by them, only the SIP signaling.
For the Asterisk can do voice recording, for example, it is necessary for the RTP traffic, or the flow of voice, go for it.
I do not use the service for recording voice or voicemail, so I do not want the RTP traffic pass by my server.
Is there a way I configure my Asterisk server this way?
Thank you so much!
canreinvite=yes
This may route through Asterisk briefly before it actually re-invites the RTP path out. There is another option that will start with Asterisk bypassed, but I think that may still be under development.
Note that you mustn’t cause Asterisk to answer until the call is bridged, and mustn’t let it generate in band progress, if you want to completely avoid RTP going through it.
[quote=“david55”]canreinvite=yes
This may route through Asterisk briefly before it actually re-invites the RTP path out. There is another option that will start with Asterisk bypassed, but I think that may still be under development.
Note that you mustn’t cause Asterisk to answer until the call is bridged, and mustn’t let it generate in band progress, if you want to completely avoid RTP going through it.[/quote]
Boy, you’re the greatest! So I love you! 
Well, I just could not understand very well his last sentence.
“Note that you mustn’t cause Asterisk to answer until the call is bridged, and mustn’t let it generate in band progress, if you want to completely avoid RTP going through it.”
If you can do me the kindness to explain better, I would be immensely grateful!
Thank you so much! 
If you use the Answer application, or generate any voice responses, Asterisk will need to set up RTP to itself. Also if you do anything that requires Asterisk to generate tones, rather than sending a SIP response that causes them to be generated up-stream, it will need an RTP connection to itself.
Oh yeah!!! I love you so much! You are the greatest!!! 
:D:D
Thank you!!!