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<DIV><FONT face=Arial size=2>Thank you Francois, that was my thoughts
also, I will experiment and post my findings. I just thought if
someone else had pursued this, I might save some time. This feature would
put the freeswitch in the same league as some of the major products
such as the Metaswitch which does support this, although they do it more with a
notify message and putting the trunk group out of service if there is no
response, I actually like that better. </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Darcy</FONT></DIV>
<BLOCKQUOTE dir=ltr
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=fdelawarde@wirelessmundi.com
href="mailto:fdelawarde@wirelessmundi.com">François</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Saturday, November 09, 2013 5:19
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] call
timeout and failover</DIV>
<DIV><BR></DIV>I see, well it's the SIP standard that defines this type of
behavior (timeout and retransmissions when not receiving a
response).<BR><BR>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>
<BLOCKQUOTE TYPE="CITE"> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><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> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE"><FONT size=2>Darcy</FONT> </BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE>----- Original Message ----- </BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE><B>From:</B> <A
href="mailto:fdelawarde@wirelessmundi.com">François</A>
</BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE><B>To:</B> <A
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users
Help</A> </BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE><B>Sent:</B> Friday, November 08, 2013 3:54 PM
</BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE><B>Subject:</B> Re: [Freeswitch-users] call timeout and
failover </BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE><BR><BR></BLOCKQUOTE></BLOCKQUOTE>
<BLOCKQUOTE TYPE="CITE">
<BLOCKQUOTE>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: <BR>
<BLOCKQUOTE TYPE="CITE"><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><BR><BR><FONT size=2>Darcy Primrose</FONT>
<BR><FONT size=2>voice2net</FONT> <PRE>_________________________________________________________________________
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Database: 3615/6799 - Release Date: 11/01/13<BR><BR></BLOCKQUOTE><PRE>_________________________________________________________________________
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<P class="" align=left color="#000000" avgcert??>No virus found in this
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href="http://www.avg.com">www.avg.com</A><BR>Version: 2014.0.4158 / Virus
Database: 3615/6799 - Release Date: 11/01/13</P></BLOCKQUOTE></BODY></HTML>