Have recently installed asterisknow, have six endpoints registered and able to make and receive calls. While analysing the sip calls with wireshark I noticed the audio streams are running from the endpoint to the asterisk now server instead of directly between the endpoints. This is not the case with other VoIP systems I have worked with. Is this normal operation for this product.
This the wrong forum for questions specifically about AsteriskNOW. The underlying Asterisk will do what you want if you set canreinvite=yes against both SIP parties, and don’t do anything (e.g use T, t, etc., Dial application options, or record calls) that requires Asterisk to monitor the RTP stream. Things get complicated if you also have NAT.
Many thanks for your help. Appologies for using the wrong forum. First posting did not scroll far enough down the main page. Found the correct forum.