Ratio of answered calls to dialed calls

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Ratio of answered calls to dialed calls

Postby garybautista » Thu Mar 20, 2008 8:48 pm

Hello!

The ratio of dialed numbers to answered calls are just ridiculously unbelievable.

CALLS TODAY: 31386
DROPPED / ANSWERED : 1 / 1198
11 AGENTS
DIAL LEVEL 4
2 E1 SIP TRUNK through GAFACHI
2 ASTERISK SERVERS
1 VICIDIAL SERVER

When I dial the numbers manually it goes through fine, and somebody picks up. So I'm assuming that it has to get passed on to the agent when it's on the dialer.

I have been playing with the DIAL TIMEOUT. I've set it up from as low as 10 sec to as high as 200 sec. I actually get more calls if it is set up at 200 sec.

When we load the same list to our non vicidial dialer it gets a lot of answered calls so it is not the list.

Any Ideas?
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Postby mflorell » Fri Mar 21, 2008 12:55 am

astGUIclient version?

Asterisk version?

counts of statuses in the list?
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Postby garybautista » Fri Mar 21, 2008 5:11 am

Vicidial Version: 2.0.4-119 BUILD: 71125-1751
Asterisk Version: 1.2.24

Example of one of the lists:
COUNTS WITHIN THIS LIST:
SUBTOTAL
A 434
B 23
CALLBK 299
DC 65
DNC 16
DROP 204
N 13
NA 3614
NEW 27
NI 54
SALE 7
TOTAL 4756
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Postby mflorell » Fri Mar 21, 2008 8:32 am

There are some bugs in the DROP call dispositioning in the 2.0.4 release. I have fixed many of these in the SVN codebase. If you get a chance, try to download the 2.0.4 branch from SVN and see if that fixes your issues.
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Postby garybautista » Fri Mar 21, 2008 10:40 am

Thanks Matt! I'll do that and let you know.
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Postby pylinuxian » Fri Mar 21, 2008 12:09 pm

is this also true for version 2.0.129 ??
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Postby mflorell » Fri Mar 21, 2008 2:23 pm

I have not tested the bugs with 2.0.3 but I would imagine it is.
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Postby garybautista » Sat Mar 22, 2008 3:13 pm

I installed the SVN Branch 2.04. I don't think it made any difference. We switch lists with respect to the time of the day. So far this is the result.

Example of one of the lists:

A 215
B 19
CALLBK 196
DC 39
DNC 11
DROP 109
INCALL 3
N 14
NA 2285
NEW 1695
NI 53
SALE 4

DIAL LEVEL: 3.5
TRUNK SHORT/FILL: 22 / 22
FILTER: NONE
TIME: 2008-03-22 12:48:32
DIALABLE LEADS: 2640
CALLS TODAY: 24598
AVG AGENTS: 9
DIAL METHOD: RATIO
HOPPER LEVEL: 200
DROPPED / ANSWERED: 0 / 413
DL DIFF: 6.55
STATUSES: NEW
LEADS IN HOPPER: 459
DROPPED PERCENT: 0%
DIFF: 72.78%
ORDER: DOWN COUNT

Any other suggestions?
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Location: Burbank, CA-USA

Postby mflorell » Sat Mar 22, 2008 6:07 pm

Did you restart all of the scripts after your install?

Do you have a single DeadAGI line on the 'h' exten in your dialplan or do you have two lines?
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Postby garybautista » Mon Mar 24, 2008 3:29 am

I rebooted the servers. So my assumption is it restarted the scripts too. My

set up is

Server 1:
MySQL
Apache
Vicidial

Server 2 and 3:
Asterisk Servers

Here's the fast AGI line in my extensions.conf on both Asterisk Server 1 and Server 2.

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten => h,2,DeadAGI(agi://127.0.0.1:4577/VD_hangup--HVcause ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}))
; Standard AGI for VICIDIAL/astGUIclient call logging
;exten => h,1,DeadAGI(call_log.agi,${EXTEN}) ; DeadAGI is new
;exten => h,2,DeadAGI(VD_hangup.agi,PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
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Postby mflorell » Mon Mar 24, 2008 7:06 am

Well, that's part of your problem, you have the old two lines in the 'h' exten. Take a look at the UPGRADE document or the extensions.conf.sample to see what the one 'h' line should be.
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Postby garybautista » Wed Mar 26, 2008 10:31 pm

I changed the settings on both asterisk servers according to the UPGRADE document. But it didn't fix the problem. I called gafachi which is my SIP provider and he told me that I have used up my lines several times and had 5+ dropped calls at those particular times.

This is weird because Gafachi allocated me with 72 channels and my maximum dial ratio is only 3.0 with 11 agents. Doesn't this only dial 33 lines at a time?

I also have 2 E1 lines. Is this enough for the number of calls I make?

I am still getting a lot of NA's and wait time averages from 1 - 1.5 minutes on a 10 hour shift. I've changed the settings of dial timeout from 15 secs to 200 secs. It seems that I get more calls faster when I set it to 200.

Any ideas?
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Location: Burbank, CA-USA

Postby mflorell » Wed Mar 26, 2008 10:59 pm

What is your max vicidial trunks set to for your server?

Have you tried a different carrier?
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Postby garybautista » Thu Mar 27, 2008 4:00 am

Dialer 1 is set to 10
Dialer 2 is set to 25

All agents are logged in on Dialer 1

I tried IAX2 with voipjet last Monday, same results.
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Postby mflorell » Thu Mar 27, 2008 7:02 am

What is the NA % listed for one day in the VDAD report?
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Postby garybautista » Thu Mar 27, 2008 10:42 am

---------- TOTALS
Total Calls placed from this Campaign: 23252
Average Call Length for all Calls in seconds: 3.28

---------- DROPS
Total DROP Calls: 9 0.04%
Percent of DROP Calls taken out of Answers: 9 / 3042 0.3%
Average Length for DROP Calls in seconds: 7.22

---------- AUTO-DIAL NO ANSWERS
Total NA calls -Busy,Disconnect,RingNoAnswer: 20210 86.92%

Average Call Length for NA Calls in seconds: 1.01
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Location: Burbank, CA-USA

Postby mflorell » Thu Mar 27, 2008 12:17 pm

I'm familiar with lists like that, you may want to try to boost the number of vicidial max trunks that you have. Dialing lists that have very low Answer percentage always requires more lines per agent to get a good agent efficiency.
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Postby garybautista » Thu Mar 27, 2008 3:26 pm

My Setup:

Apache/MySQL
Asterisk1
Asterisk2

All Agents logged in to Asterisk 1

I have 3 Questions.

1. Can I have half of the agents log in to asterisk2?
2. Will this improve the servers performance?
3. What is the maximum simultaneous calls I can have with 2 E1 lines using ulaw?
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Location: Burbank, CA-USA

Postby mflorell » Fri Mar 28, 2008 1:26 pm

At this point what server the agents are on is not a factor if you are using balance auto dialing.

2 x E1s = 64 RBS voice channels. If you use ULAW VOIP you actually loose capacity and will probably only be able to get 50-55 channel reliably.
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