Hi,<br><br>2 things to look at...<br><br>1. Under your dial-peer look into the "<b class="cBold">monitor probe icmp-ping</b><span class="content"></span> X.X.X.X" command<br>If destination does not respond it shuts down that dial-peer.. therefore failover should be instant.<br>
<br>2. Global config - you can adjust your timers - it will try 6 times, with 3 sec timeout by default<br>sip-ua<br>retry invite 3<br>timers trying 2<br><br>Use combo of trying time and number of invites to reach your goal.<br>
<br>Cheers,<br><br>Tim<br><br><div class="gmail_quote">On Mon, Jul 13, 2009 at 6:10 PM, Gavrilov, Anatoly <span dir="ltr"><<a href="mailto:Anatoly.Gavrilov@gsjbw.com">Anatoly.Gavrilov@gsjbw.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-AU">
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<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">Hi Guys, </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">I’ve configured two simple dial-peers on a voice
gateway:</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">dial-peer voice 1111 voip</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> description test Primary</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> preference 1</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> destination-pattern 1199</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session protocol sipv2</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session target ipv4:10.10.10.10</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session transport udp</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">dial-peer voice 1112 voip</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> description test Primary</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> preference 2</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> destination-pattern 1199</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session protocol sipv2</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session target ipv4:10.10.10.11</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> session transport udp</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">I’ve tested hunting and I’ve found the following
behaviour:</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">If 10.10.10.10 is pingable, the call immediately goes to the
second dial-peer</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">But if 10.10.10.10 is not pingable, Voice gateway will wait
60 seconds before the call goes to the second dial-peer. </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">Do you know what command I can use to reduce that value to 5
seconds? Like “h225 timeout tcp establish 5” but for the SIP
calls? </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">Thanks and Cheers!</span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;"> </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">Best Regards, </span></font></p>
<p><font size="2" face="Arial"><span style="font-size: 10pt; font-family: Arial;">Anatoly</span></font></p>
<p><font face="Verdana"><span style="font-family: Verdana;"> </span></font></p>
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