check in : http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite<br> <h1>Asterisk sip.conf, peer definition: canreinvite option </h1> <br> This peer option in <b>sip.conf</b> is used to tell the <a title="" href="http://www.voip-info.org/wiki/view/Asterisk">Asterisk</a> server to never issue a reinvite to the client. This is used to interoperate with some (buggy) hardware that crashes if we reinvite, such as the common Cisco ATA 186. <br> <br> When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other. <br> <br> <ul> <li> If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all. </li><li> If the
clients use different codecs, Asterisk will not issue a re-invite. </li><li> If the Dial() command contains <i>t</i>, ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite. </li> </ul> <br> <b>'canreinvite=no'</b> stops the sending of the (re)INVITEs once the call is established. From messages in the archives and the Asterisk handbook one finds out that the Cisco ATA-186 does not handle the (re)INVITE well. This is necessary if the client and the Asterisk server is on opposite sides of a <a title="" href="http://www.voip-info.org/wiki/index.php?page=NAT+and+VOIP">NAT</a> gateway or firewall. <br> <br> <h2>Notes </h2> <ul> <li> <b>reinvite=yes/no is plain wrong</b>, even if you see it mentioned in example .conf files. The correct syntax is <b>can</b>reinvite=yes/no </li><li> Connecting media paths direct to an endpoint behind NAT won't be pretty. Especially if both devices are behind NAT. You might want to try using SER's
nathelper in conjunction since nathelper.so can rewrite the SDP so that the private IP addresses are not included in the re-invite. ....<br> </li> </ul> <br><br><b><i>support <brian@ezzitel.com></i></b> a écrit :<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> So you want asterisk not to proxy the media.. Simple. Put<br>canreinvite=yes in your peer in sip.conf. It will do it automatically.<br><br><br><br>On Fri, 2006-02-17 at 01:19 +1100, Zafer Khodr wrote:<br>> I am trying to setup asterisk as shown below.<br>> <br>> <br>> <br>> xxx.xxx.xxx.xxx<br>> /=====(Call Origination)====\<br>> / | \<br>> R / |S \<br>> T / |I \ R<br>> P / |P \ T<br>> / | \ P<br>> S / Asterisk \ <br>>
T / /\ \ S<br>> R / / \ \ T<br>> E / / \ \ R<br>> A / / \ \ E<br>> M / / \ \ A<br>> / / \ \ M<br>> / / \ \<br>> / / \ \<br>> / __________/ \__________ \<br>> / | | \<br>> / | | \<br>> yyy.yyy.yyy.yyy zzz.zzz.zzz.zzz<br>> Terminating Gateway 1 Terminating Gateway 2<br>> <br>> <br>> <br>> The main objective to to have asterisk in the path of the call but the RTP<br>> to go directly between the originating and terminating IP's.<br>> I have had a play around with canreinvite but doesn't
seem to make a<br>> difference.<br>> <br>> If someone could please help me out by pointing me in the right direction,<br>> that would be great<br>> <br>> Regards,<br>> Zafer<br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> _______________________________________________<br>> --Bandwidth and Colocation provided by Easynews.com --<br>> <br>> asterisk-biz mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-biz<br>_______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-biz mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-biz<br></blockquote><br><p>
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