The variable is <b>canreinvite.</b><div><b><span style="font-weight:normal">Please check on voipinfo. If canreinvite is enabled then only SIP signaling is passed through Asterisk and the media is not passed through Asterisk resulting in less bandwidth usage and probably less jitter buffer, etc,,,,if you are two phones are closer to each other than a round trip to Asterisk server.</span></b></div>
<div><b><span style="font-weight:normal"><br></span></b></div><div><b><span style="font-weight:normal">On the flip side, you can't record these calls because no media is sent through Asterisk.</span></b></div><div><b><span style="font-weight:normal"><br>
</span></b></div><div><b><span style="font-weight:normal">-Bruce<br></span></b><br><div class="gmail_quote">On Fri, Jul 9, 2010 at 1:48 PM, <span dir="ltr"><<a href="mailto:unserossi@aol.com" target="_blank">unserossi@aol.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><font color="black" size="2" face="arial"><font size="2"><font face="Arial, Helvetica, sans-serif">Hi all,<br>
<br>
i have a beginners question. How are SIP calls and IAX2 calls processed by Asterisk over the network?<br>
What i mean is, is there a permanent connection required between the Asterisk Server and the clients or is the Asterisk Server only involved for lets call it the "routing"?<br>
<br>