I guess it has to be on the Trunk and one of the either user or peer and the opposing party shouldn't have it as no.<div><br></div><div>But, to full proof urself, put it on the trunk and both users. Basically put it anywhere that takes it.</div>
<div><br></div><div><a href="http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite">http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite</a></div><div><br></div><div><a href="http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite"></a>-Bruce<br>
<br><div class="gmail_quote">On Fri, Jul 9, 2010 at 2:40 PM, <span dir="ltr"><<a href="mailto:unserossi@aol.com">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">Sou<font color="black" face="arial" size="2">nds great, thanks for your answer.<br>
Do i need to set this on the trunk, the friend or on both?
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<div style="font-family:arial,helvetica;font-size:10pt;color:black"><div><div></div><div class="h5">-----Original Message-----<br>
From: bruce bruce <<a href="mailto:bruceb444@gmail.com" target="_blank">bruceb444@gmail.com</a>><br>
To: Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
Sent: Fri, Jul 9, 2010 8:13 pm<br>
Subject: Re: [asterisk-users] General network question regarding SIP and IAX2<br>
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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>
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<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>
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<div><b><span style="font-weight:normal">-Bruce<br>
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<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:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><font color="black" face="arial" size="2"><font size="2"><font face="Arial, Helvetica, sans-serif">Hi all,<br>
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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>
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