<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I think you wanna use progress_timeout <div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout">http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout</a></div><div><br></div><div>/b</div><div><br><div><div>On Aug 18, 2009, at 10:24 AM, Hristo Trendev wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">I am trying to implement failover dialing plan as described in:<br><a href="http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout">http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout</a><br><br>I figured out that originate_timeout must be passed as<br>{originate_timeout=<timeout>} in front of the dial string to have any<br>effect (setting it as channel variable as described in the example<br>above has no effect).<br><br>I have set the timeout to 1 second, so expected behavior is to try the<br>second gateway if no response is received from the first one in 1<br>second. The problem is that FS cancels the first request with<br>[NO_ANSWER] and tries to route the call via the second gateway even<br>though it receives response from the first during that 1 second.<br><br>The response received is "100 Trying" provisional response (checked<br>with sofia siptrace). I'm guessing that either 100 provisional<br>responses don't cancel the originate_timeout timer (bug?) or I am<br>doing it the wrong way.<br><br>I was also thinking of using the timer-T1 or timer-T1X64 parameter in<br>the sip profile, but I need this to be set per dial string, not per<br>profile, besides, it seems that these timers (T1, T1X64) affect both<br>invite and non-invite requests, so this is not really an option.<br>Also, I tried leg_timeout, but it doesn't really do what I need it to.<br><br>Anyone has any idea how to implement this?<br></blockquote></div><br></div></body></html>