[Freeswitch-users] How long did the phone ring?

Fraser Redmond fraserredmond at gmail.com
Wed Mar 17 10:17:54 PDT 2010


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 at 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 at 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.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100317/5530c413/attachment-0002.html 


More information about the FreeSWITCH-users mailing list