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I see, well it's the SIP standard that defines this type of behavior (timeout and retransmissions when not receiving a response).<BR>
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It might be possible to do what you want by tweaking the SIP timers (sofia profile param "timer-T1", "timer-T2", ...). Otherwise you would probably need to modify the sofia stack.<BR>
<BR>
François.<BR>
<BR>
On Fri, 2013-11-08 at 22:51 -0500, fs wrote:<BR>
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<FONT SIZE="2">Yes I did, but again, that times out on a progress message, if you have a dead carrier you are sending the message to, it waits for the leg timeout which in our case needs to be set to about 12 seconds to allow for cell phones etc. So it takes quite a while to switch, the calling party usually hangs up or the call completion time is just to long.</FONT>
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<FONT SIZE="2">Darcy</FONT>
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<B>From:</B> <A HREF="mailto:fdelawarde@wirelessmundi.com">François</A>
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<B>To:</B> <A HREF="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
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<B>Sent:</B> Friday, November 08, 2013 3:54 PM
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<B>Subject:</B> Re: [Freeswitch-users] call timeout and failover
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Did you try with the "leg_progress_timeout" variable?<BR>
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François.<BR>
<BR>
On Fri, 2013-11-08 at 15:06 -0500, fs wrote: <BR>
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<FONT SIZE="2">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. </FONT><BR>
<FONT SIZE="2">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.</FONT> <BR>
<FONT SIZE="2">Also, when calling internationally, longer times are quite common.</FONT> <BR>
<FONT SIZE="2">Perhaps this is already implemented but I have not been able to find it.</FONT> <BR>
<FONT SIZE="2">449.776291 SIP/SDP Request: INVITE sip:1613802xxxx@xxx.xxx.xxx.xx, with session description</FONT><BR>
<FONT SIZE="2">449.782429 SIP Status: 100 Trying</FONT><BR>
<FONT SIZE="2">456.211562 SIP/SDP Status: 183 Session Progress, with session description</FONT><BR>
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<BR>
<FONT SIZE="2">Darcy Primrose</FONT> <BR>
<FONT SIZE="2">voice2net</FONT>
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