vicidial.org

Hi,

I am experiencing problems in that the dialer,it is dialing quite slowly. I'm on ADAPT-TAPERED.
I have two people and it is only dialing one line at a time.

if you have less than 50 agents you should not be on any of the adapt settings. the "predictive" mode of a dialer requires a lot of data to determine the best settings. with two agents there will never be enough data on a shift to make it work.

use "ratio".

Thanks, what is the best ratio level for 1-3 agents?

I try to put it at 2 but there is a lot of dropped calls (10%+)

It changes from person to person, campaign to campaign, list to list and time of day. The best setting is the one that results in 3% or less of dropped calls. But that's not the only setting!

You also need to be sure that "Available Only Tally" is set to "Y".

Set your Admin->Servers->Max Calls Per Second to "1".

Thanks,

Is it possible to pass about-to-be-dropped calls to a less active agent?

Me I'm just coding on the side, I don't want to do telemarketing but I'd rather have it go to me then just be dropped.

Yes, as a matter of fact it is. You can set the drop call seconds to a nice LOW number (like ... Zero?) and drop those calls to an ingroup with a group of less active (but available and logged in!) agents/managers who can then catch those calls and quickly dispose of them if they are not "sale!", which will reduce your dropped call percentage legally (live agents!) and perhaps even generate a few extra sales.

Nice, will do.

Ok, I did just that but for some reason I am getting a lot of calls where I hear nothing

test it. likely clients hanging up. that's what happens when you "transfer" someone you called. (I hang up when that happens, don't you?)

That's what I taught...

I'm just gonna transfer directly to the softphone/extension because I assigned myself to an ingroup but this seems awkward/too slow.

EDIT: Nvm, there's no option to transfer directly to a softphone instead of an ingroup

Ok how do I work around that so that the transition is seemless? There must be a setting I'm not doing right.

transfer to an ingroup anyway. the controls in the ingroups are very robust. there are settings for what to do if there is no agent available in the ingroup or during after hours. these allow bounces to various locations instantly based on the scenario.

Ok then there's something I don't understand, I'm the only one currently in my in-group,I'm not doing any inbound calls, but it still takes some time for the call to get there.

When i put extention, it rings automatically, how come it is not the case there?

There must be a decision-making process involved that is not needed in the extension transfer (after all, there's only one extension, no decision).

Ordinarily the only time I experience any timing issues is when the server is seriously overloaded or it's a "virtual" system. Otherwise, there are no timing delays a human will notice.

If there is I'm truly not sure what it is, In the campaign, the drop call seconds is set to 0, and inbound group is set to 201 -agent201.

Right now I'm working late in the office when all my agents are gone. When I go to the individual recordings, i hear nothing. When I call directly my DID number and dial the 201 extention (which also points to the 201 ingroup) it seems to be near instantaneous.

Ok its not instantaneous, there's a 2-3 seconds delay.

watch your asterisk command line. use agi debug and sip debug to track what happens during your test.

I just have no real clue as to what is happening underneath the hood, I can make some sense out of the SIP requests and the AGI requests, but I have no idea what is reffering to what and where does the requests go. If you can give me a clue as to what to look for specifically, that would be appreciated even if I'm still working on it. In my test, it seems to go near instantaneously (2-3 second) but when I'm doing it live, people are waiting for much longer then this

I know the RIGHT answer would be to go dig in some Asterisk manual and find out what every command means(or at least have a rough enough idea to make an educated guess), but I don't have tons of time to fix this really unfortunately.

Ok, is there any kind of ID I can use to track what is going where??

EDIT: OK , call-id in the SIP requests looks about right

I have 3 computers right now , I used

1- Outbound agent
2- Inbound agent
3- Not used

OK, if I understand correctly, it is retransmitting NAT a couple of time, it sends the call to agent #3(???) and then transmits it to agent #2...

Ok, I just started calling Boston and it seems to be working as if by magic, is it possible that the issue is external?

Ok nvm I got my hopes up for nothing

during a debug session, you can use several tricks.

sip set debug ip xx.xx.xx.xx

asterisk -rx "logger rotate" (and modify /etc/asterisk/logger.conf to determine what entries go in which files). You can rotate the log, create your test, rotate again and you'll end up with a single file that has your test time in it that you may peruse at will. That file will no longer be modified. Of course the logs are in /var/log/asterisk

ok thaanks Ill play around with it