[Asterisk-Users] I Don't Want Asterisk in the Media Path

[Eric Wieling aka ManxPower]

Dhennys Pestana wrote:
> I'm trying to find a way to connect two (or more) extensions directly without
> being kept in the middle during the conversation but it won't happen.

Asterisk will always stay in the SIP signaling path.  It can get out of 
the RTP path (only way to really see this is using something like 
tcpdump since sip show channels shows the signaling not the RTP path). 
Asterisk CANNOT get out of the RTP path if you are using the "t" or "T" 
option to dial (maybe other options too) or if the codec for the two 
legs of the call are different.