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