Yeah, that's what I had expected, and was why I was confused.

I've done some more tests, and tracked the results a little closer in a spreadsheet, and it seems that when I call the extension directly from another sipphone the  Caller-Channel-Answered-Time minus Caller-Channel-Created-Time  matches the ring time, so that scenario is fine.

When I call into an IVR with the javascript dialplan and then create the new session and bridge them that way, the Answered-Time and Created-Time, that are reported after the call ends are reported on the A-leg's Created & Answered.

The good news is that the Progress-Time is reported on when the B-leg started ringing, so I can know when the call started ringing.
The bad news is that the Progress-Media-Time is always blank

I can take the Progress-Time and compare it to the system-clock, and that should generally be accurate to a second or so, which is better than nothing, but it'd be nice to do it properly.

Any other ideas? Is there any channel-variables I can set that would be worth playing with? I'm currently only setting:
ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true

Cheers,
Fraser




On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
even better you have the progress and progress_media timestamps too
so you can measure from the instance you got the first ringing indication
so then you can also measure how long it took to start ringing too (PDD)


On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian@freeswitch.org> wrote:
Call start answer time minus call start time = ring time in the CDR

/b

On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:

> I'm converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I've worked out how to do nearly everything, except for tracking one important metric: How long the phone rang before an agent picked it up.





--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org