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

Wed Mar 17 12:46:48 PDT 2010

SIP 180 = your phone rings (Ringing message)
SIP 183 + sdp = early media or inband ringing (Session Progress message)

You have both vars to watch for.

Ognjen

On Wed, Mar 17, 2010 at 8:18 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> Sorry - newbie question: what is the 180 or 183? Can I manipulate that any
> way?
>
> I've realised since my last email that even having the progress_time
> doesn't help without the progress_media_time or some way to find when the
> ringing stopped - getting the server timestamp wouldn't happen until the
> call ended, at which point I can't split the ring time from the talk time.
>
> Surely there can't be something that Asterisk handles well that FreeSwitch
> doesn't? ;-)
>
> I'm looking into the CDR now - might have to parse the log files to get the
> end time from there :-(
>
> Cheers,
> Fraser
>
>
>
>
>
> On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> progress_media_time is only set if you get a 183
>> if you get a 180 with no media it uses progress_time instead
>>
>> as long as you have one or the other you have the timestamp of when it
>> started ringing.
>>
>>
>>
>>
>>
>>
>> On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond at gmail.com
>> > wrote:
>>
>>> 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
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>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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>
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>>>>
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>>>
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>>>
>>
>>
>> --
>> 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
>>
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>>
>
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