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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Hi,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Also try with setting the ignore_early_media=true or =ring_ready channel variable or in the dial string.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><a href="http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media">http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media</a>
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><a href="http://wiki.freeswitch.org/wiki/Early_Media#Ignoring_Early_Media">http://wiki.freeswitch.org/wiki/Early_Media#Ignoring_Early_Media</a><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Robert<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
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<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""> François [mailto:fdelawarde@wirelessmundi.com]
<br>
<b>Sent:</b> Friday, November 08, 2013 12:54 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] call timeout and failover<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Did you try with the "leg_progress_timeout" variable?<br>
<br>
François.<br>
<br>
On Fri, 2013-11-08 at 15:06 -0500, fs wrote: <o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:10.0pt">I wonder if anyone has any thoughts on this. Using continue on fail works fine for failover but to the best of my knowledge it only works if you do not get an answer or session progress. The trouble with
this is if you are calling cell phones etc, it can take up to 12 seconds to get the messages. I would like to explore if failover could be instituted by not receiving a Trying message. This is usually returned immediately and does stop the freeswitch from
sending more invites. That way, a 4 second timer could be used and switch to an alternate route early and still give reasonable completion times.
</span><o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">The tcap trace below shows a typical call to a cell in our area, the trying is almost immediate while the cell in this case took 5.5 seconds. I have seen this quite frequently take as long as 12 seconds.</span>
<o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Also, when calling internationally, longer times are quite common.</span>
<o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Perhaps this is already implemented but I have not been able to find it.</span>
<o:p></o:p></p>
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<p class="MsoNormal"> <o:p></o:p></p>
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<p class="MsoNormal" style="margin-bottom:12.0pt"><span style="font-size:10.0pt">449.776291 SIP/SDP Request: INVITE
<a href="sip:1613802xxxx@xxx.xxx.xxx.xx">sip:1613802xxxx@xxx.xxx.xxx.xx</a>, with session description</span><br>
<span style="font-size:10.0pt">449.782429 SIP Status: 100 Trying</span><br>
<span style="font-size:10.0pt">456.211562 SIP/SDP Status: 183 Session Progress, with session description</span><br>
<br>
<o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">Darcy Primrose</span> <o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:10.0pt">voice2net</span> <o:p></o:p></p>
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<pre><o:p> </o:p></pre>
<pre>_________________________________________________________________________<o:p></o:p></pre>
<pre>Professional FreeSWITCH Consulting Services:<o:p></o:p></pre>
<pre><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><o:p></o:p></pre>
<pre><a href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<o:p></o:p></pre>
<pre><a href="http://www.cudatel.com">http://www.cudatel.com</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>Official FreeSWITCH Sites<o:p></o:p></pre>
<pre><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><o:p></o:p></pre>
<pre><a href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a><o:p></o:p></pre>
<pre><a href="http://www.cluecon.com">http://www.cluecon.com</a><o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>FreeSWITCH-users mailing list<o:p></o:p></pre>
<pre><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><o:p></o:p></pre>
<pre><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><o:p></o:p></pre>
<pre>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><o:p></o:p></pre>
<pre><a href="http://www.freeswitch.org">http://www.freeswitch.org</a><o:p></o:p></pre>
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<p class="MsoNormal"><o:p> </o:p></p>
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