How long did the phone ring?
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. I'm not even entirely sure what to call it in FreeSwitch lingo - so that would be a good starting point :-) (Media-duration maybe?) I can work out the length of ringing for calls that didn't connect, by logging when the bridge is attempted, and after it fails and comparing the time. But if the person being called picks up, then all I know from that is the total length of the call including the ringing/media time, not the media time and connected time as 2 separate values The info application gives several useful looking variables like: Caller-Channel-Progress-Media-Time Caller-Channel-Transfer-Time ...but they're still set to zero. I thought I might be able to subtract Caller-Channel-Created-Time from Caller-Channel-Answered-Time to get the milliseconds of how long the phone rang for - but there's no reliable correlation between the difference and how long the phone rang. In asterisk I achieved this by logging when I was about to make a call, then running a macro when the Dial() connected, like so: exten => s,n,Macro(Dial_Attempted) exten => s,n,Dial(SIP/1001,20,M(Dial_Succeeded)) exten => s,n,Macro(Dial_Failed) ;Only gets here if the Dial didn't connect. Is there any way to do this in FreeSwitch? Cheers, Fraser
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.
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.
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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.
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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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@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@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.
_______________________________________________ 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
-- Anthony Minessale II
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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@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@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@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.
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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i'll try again progress_media and progress are both the same for your purposes you will only have exactly one of the 2 set. its the exact time it started ringing. for our sake, if we have either of them we will take the one that is not zero and call it the progress time. so you have the created time, the progress time , the answered time and the hangup time you always have created time and hangup time but not always progress and answered so talk time is (hangup_time - (answered_time unless its 0)) or if answered_time is 0 there is no talk time ring time is: (answered time unless its zero otherwise hangup_time) - created_time edit conf/autoload_configs/cdr_csv.conf.xml uncomment: <param name="debug" value="true"/> and restart FS now when you hangup all your calls you will see the values you have to work with in the cdr if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs. On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond@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@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@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@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.
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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Thanks, I suspect this is what I really needed:
if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs.
I guess there's more for me to un-learn from Asterisk than I'd thought :-( Cheers, Fraser On Wed, Mar 17, 2010 at 7:41 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
i'll try again
progress_media and progress are both the same for your purposes you will only have exactly one of the 2 set. its the exact time it started ringing.
for our sake, if we have either of them we will take the one that is not zero and call it the progress time.
so you have the created time, the progress time , the answered time and the hangup time
you always have created time and hangup time but not always progress and answered
so talk time is (hangup_time - (answered_time unless its 0)) or if answered_time is 0 there is no talk time ring time is: (answered time unless its zero otherwise hangup_time) - created_time
edit conf/autoload_configs/cdr_csv.conf.xml
uncomment:
<param name="debug" value="true"/>
and restart FS
now when you hangup all your calls you will see the values you have to work with in the cdr
if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs.
On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond@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@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@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@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.
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On Wed, Mar 17, 2010 at 1:10 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
Thanks, I suspect this is what I really needed:
if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs.
I guess there's more for me to un-learn from Asterisk than I'd thought :-(
Yes, there probably is. FreeSWITCH is definitely a paradigm shift, but once you realize the power under the hood you'll never go back to Asterisk. If you *really* want to get fancy you can use mod_xml_cdr and get a true 3D picture of your call. You can even use it to post the XML CDRs to an external web server and do all your billing offline. :D You might want to join #freeswitch on irc.freenode.net and rub elbows with the other Asterisk refugees who've made the transition. -MC
On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc@freeswitch.org> wrote:
If you *really* want to get fancy you can use mod_xml_cdr and get a true 3D picture of your call. You can even use it to post the XML CDRs to an external web server and do all your billing offline. :D
Yes, I was planning on using mod_xml_cdr to pass the info to our webserver to update the database, once I know that mod_cdr_csv is getting the right data.... which it still isn't :-( Much the same problem as before - in mod_cdr_csv's logs the start-time and answered-time aren't what I expect. I'm gradually getting a better handle on whats happening... The 4 significant events in my dialplan are: ivr-started (I dial a switchboard number, and enter the extn I want to be connected to) call-started (I'm connected to that extn) call-ended (the call with the extn ends) ivr-ended (I may return to the extn, or end at the same time as the call) Master.csv contains 2 rows, both with from and to numbers matching the caller and callee appropriately. It seems like the first/blind_transfer row is not picking up the info from the call that it ought to, and some of it is being recorded against the second/main ivr row and then lost when it gets new data. First row has: start = call-started answer = empty end = call-started (probably actually call-connected) hangup_cause = BLIND_TRANSFER bleg uuid = empty Second row (also appears in the separate csv file for the caller's extension) has start = ivr-started answer = ivr-started end = ivr-ended hangup_cause = NORMAL_CLEARING bleg uuid = present Cheers, Fraser
you might want to turn on B-leg logging On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
On Wed, Mar 17, 2010 at 8:34 PM, Michael Collins <msc@freeswitch.org>wrote:
If you *really* want to get fancy you can use mod_xml_cdr and get a true 3D picture of your call. You can even use it to post the XML CDRs to an external web server and do all your billing offline. :D
Yes, I was planning on using mod_xml_cdr to pass the info to our webserver to update the database, once I know that mod_cdr_csv is getting the right data.... which it still isn't :-(
Much the same problem as before - in mod_cdr_csv's logs the start-time and answered-time aren't what I expect. I'm gradually getting a better handle on whats happening...
The 4 significant events in my dialplan are:
ivr-started (I dial a switchboard number, and enter the extn I want to be connected to) call-started (I'm connected to that extn) call-ended (the call with the extn ends) ivr-ended (I may return to the extn, or end at the same time as the call)
Master.csv contains 2 rows, both with from and to numbers matching the caller and callee appropriately.
It seems like the first/blind_transfer row is not picking up the info from the call that it ought to, and some of it is being recorded against the second/main ivr row and then lost when it gets new data.
First row has: start = call-started answer = empty end = call-started (probably actually call-connected) hangup_cause = BLIND_TRANSFER bleg uuid = empty
Second row (also appears in the separate csv file for the caller's extension) has start = ivr-started answer = ivr-started end = ivr-ended hangup_cause = NORMAL_CLEARING bleg uuid = present
Cheers, Fraser
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lol - yeah, I guess that would help! /smacks head against nearby wall/ I'm in the UK and am heading to bed now - I'll sleep much better thanks to those 8 words from you :-) Thanks a bunch! Cheers, Fraser On Wed, Mar 17, 2010 at 10:37 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
you might want to turn on B-leg logging
On Wed, Mar 17, 2010 at 5:23 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
It seems like the first/blind_transfer row is not picking up the info from the call that it ought to, and some of it is being recorded against the second/main ivr row and then lost when it gets new data.
FYI this is now on the wiki: http://wiki.freeswitch.org/wiki/CDR#Calculating_Various_Time_Values_For_A_Ca... -MC On Wed, Mar 17, 2010 at 12:41 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
i'll try again
progress_media and progress are both the same for your purposes you will only have exactly one of the 2 set. its the exact time it started ringing.
for our sake, if we have either of them we will take the one that is not zero and call it the progress time.
so you have the created time, the progress time , the answered time and the hangup time
you always have created time and hangup time but not always progress and answered
so talk time is (hangup_time - (answered_time unless its 0)) or if answered_time is 0 there is no talk time ring time is: (answered time unless its zero otherwise hangup_time) - created_time
edit conf/autoload_configs/cdr_csv.conf.xml
uncomment:
<param name="debug" value="true"/>
and restart FS
now when you hangup all your calls you will see the values you have to work with in the cdr
if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs.
On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond@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@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@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@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.
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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@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@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@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@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.
_______________________________________________ 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
-- Anthony Minessale II
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They are referring to SIP 180 and SIP 183. It's kind of a mess (it is SIP) but both are non-final response codes to indicate call progress. Typically speaking, 180 Ringing is returned without media and tells the receiving device (FreeSWITCH or your local SIP phone in this case) to activate a local ringback generator to inform the caller the remote end is progressing. 180 without media is typically used when calling SIP phones. They know that they only actually "ring". 183 is typically used with media. The media contains audio to tell the caller what is happening. Typically this audio sounds just like ringback (depending on region) but it could just as easily be a ringback tone (Britney Spears or anything else), audio from the remote end, or just about anything else the caller might need to know (even an error message). Most carriers *should* consistently give you 183 with media while the call progresses, whether the remote end is actually ringing or not depends. You have to listen to the audio :). BTW, as are most things with SIP these are generalities and many not happen in your environment, most stacks, for example, will handle 180/183 interchangeably - it's the presence of media (SDP) that matters. FreeSWITCH has a multitude of options to deal with these scenarios. First things first - what does your carrier return? On Wed, Mar 17, 2010 at 3:18 PM, Fraser Redmond <fraserredmond@gmail.com> wrote:
Sorry - newbie question: what is the 180 or 183? Can I manipulate that any way?
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com
participants (6)
-
Anthony Minessale
-
Brian West
-
Fraser Redmond
-
Kristian Kielhofner
-
Michael Collins
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Ognjen Seslija