vicidial.org

I want to make this configuration:

I want that the agent be able to dispo. the disconect numbers, and busy. I'm using call method= Radio.

What's happend is that the leads have some disconnected numbers, so the Asterisk never pass that calls to the agent, the console shows in that moment "everybody is busy/congested at this time" and when you make a the call manually from you cellphone, that number is disconnected or is busy.

Any ideas?

Goautodial 3.3
VERSION: 2.9-441a
BUILD: 140612-1628
Asterisk 1.8.23.0

If you want to pass every call to an agent, why aren't you using MANUAL or INBOUND_MAN dial methods?

The dial method must be ratio. That's how we want to work, without agents make the process to make a call. Ratio make the call without agent decision.
Did you know how to make this done? Did you have an idea?

If you are auto-dialing, vicidial will not send calls with no Answer on the far end to the agent. If there is no active call to send to an agent then there is nothing to route, so there is really no way to do what you want to do using the RATIO dial method.

You know how to dispo this calls automatically with a customs disposition? ... Because the problem here is that all calls that aren't pass throught the agent, these calls doesn't have a correct disposition, these have the dispo "NEW" and called by "VDAD"... And it cause this problem: I'm using the manual alt_number dialing, the leads have sometimes 2 alt numbers and some leads have just 1 + the primary. And when I put the primary number of these leads, and vicidial don't pass the calls to the agent, the agent can't call to the alt_numbers of that lead.

Thanks for your quickly response and your help.

I would usually suggest using MULTI_LEAD auto alt dialing, you won't have this problem if you use that method. Of course you would have to load your leads differently, but it would allow for more control.

As to why your leads are being set as NEW even after they are called, that sounds like a configuration problem.

Also, I would recommend upgrading your system to the latest svn/trunk code. The version you are using has a lot of bugs that have been fixed in the last 8 months.

That's true. If I use MULTI_LEAD auto alt dialing, it call the 3 alt number even if is present in alt_phone or address3 right? and even if the primary is disconected right? ... When I use that method, I upload the leads at the same way... With the 3th old loader of leads.

What is the correct disposition of these calls that are not passed throuth the agent? DROP or ERI or? ... Every call that have the next error in the asterisk:

[Feb 10 14:22:32] -- Executing [144924402484@default:1] NoOp("Local/144924402484@default-000002e9;2", "quita prefijo 1") in new stack
[Feb 10 14:22:32] -- Executing [144924402484@default:2] Set("Local/144924402484@default-000002e9;2", "nuevaext=44924402484") in new stack
[Feb 10 14:22:32] -- Executing [144924402484@default:3] Goto("Local/144924402484@default-000002e9;2", "outbound,44924402484,1") in new stack
[Feb 10 14:22:32] -- Goto (outbound,44924402484,1)
[Feb 10 14:22:32] -- Executing [44924402484@outbound:1] NoOp("Local/144924402484@default-000002e9;2", "") in new stack
[Feb 10 14:22:32] -- Executing [44924402484@outbound:2] Dial("Local/144924402484@default-000002e9;2", "SIP/924402484@outbound") in new stack
[Feb 10 14:22:32] == Using SIP RTP CoS mark 5
[Feb 10 14:22:32] -- Called SIP/924402484@outbound
[Feb 10 14:22:32] -- SIP/outbound-000001e0 is making progress passing it to Local/144924402484@default-000002e9;2
[Feb 10 14:22:33] -- SIP/outbound-000001e0 is ringing
[Feb 10 14:22:33] -- SIP/outbound-000001e0 is making progress passing it to Local/144924402484@default-000002e9;2
[Feb 10 14:22:37] == Everyone is busy/congested at this time (1:0/0/1) <<<<<<<<<<<<<<<<<<< HERE IS THE ERROR******
[Feb 10 14:22:37] -- Executing [44924402484@outbound:3] Hangup("Local/144924402484@default-000002e9;2", "") in new stack
[Feb 10 14:22:37] == Spawn extension (outbound, 44924402484, 3) exited non-zero on 'Local/144924402484@default-000002e9;2'
[Feb 10 14:22:37] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:23:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:23:01] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:23:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:23:02] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:23:06] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:23:06] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:23:30] > Refreshing DNS lookups.
[Feb 10 14:24:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:24:01] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:24:01] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:24:02] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 14:24:06] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 14:24:06] == Manager 'sendcron' logged off from 127.0.0.1

And before that error asterisk takes like a 10 mins with the same errror, but if you call that numbers these are disconnected.

What svn/trunk do you recommend?