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<p class=MsoNormal><span style='color:#1F497D'>I did the below with SIP
dial-peers that are pointing to another server over VPN connection “fall
back to PRI” and it worked fine<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>sip-ua <o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> retry invite 3<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> timers trying 350<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>however try to work carefully
with these timers as it depends on your environment…if you set it too short
your SIP calls may fail.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<p class=MsoNormal style='background:white'><span style='font-size:10.5pt;
color:#1F497D'> Best Regards;<o:p></o:p></span></p>
<p class=MsoNormal style='background:white'><span style='font-size:10.5pt;
color:#1F497D'> Ahmed Elnagar<o:p></o:p></span></p>
<p class=MsoNormal style='background:white'><span style='font-size:10.5pt;
color:#1F497D'> Senior Network PS Engineer<o:p></o:p></span></p>
<p class=MsoNormal style='background:white'><span style='font-size:10.5pt;
color:#1F497D'> <img width=63 height=63 id="Picture_x0020_0"
src="cid:image001.jpg@01CABC02.BC625D80" alt="ccie_voice_large.gif"><img
width=63 height=63 id="Picture_x0020_1" src="cid:image002.jpg@01CABC02.BC625D80"
alt="ccvp_voice_large.gif"><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
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<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Matthew Loraditch<br>
<b>Sent:</b> Friday, March 05, 2010 12:26 AM<br>
<b>To:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> [cisco-voip] Timeout for h323 dial peers?<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>We have sip trunks on an h323 gateway setup as a primary
path for some outbound calls with PRIs on an MGCP gateway as backup.
It seems to take a long time on the order of sometimes a minute or longer
for a call to decide it won’t complete as SIP and then failover to
the PRI. Is there a way to decrease this period? If a call is going to connect
on SIP it does so fairly quickly I’d say less than 20 seconds and
probably less than 10.<o:p></o:p></p>
<p class=MsoNormal>Thanks!<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><b><span style='color:black'>Matthew Loraditch, CCNA</span></b><span
style='color:black'><br>
1965 Greenspring Drive<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'>Timonium, MD 21093 <br>
<a href="mailto:support@heliontechnologies.com"><span style='color:black'>support@heliontechnologies.com</span></a><br>
(p) (410) 252-8830<br>
(F) (443) 541-1593<br>
<br>
Visit us at <a href="http://www.heliontechnologies.com/"><span
style='color:black'>www.heliontechnologies.com</span></a> <br>
Support Issue? Email <a href="mailto:support@heliontechnologies.com"><span
style='color:black'>support@heliontechnologies.com</span></a> for fast assistance!</span><o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
</div>
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