Ration dialing not working properly

[muyousif]

Hi experts, currently running vicidial cluster with 200 agents and dialing method is ratio. I have set ratio dialing to 2. Most of the time my agents are waiting too long for calls, Vicidial shows me a large number of calls are being dialed/ringing but I see very low number of calls are reaching to my billing/fail over server which has sip trunks with multiple carriers. If I increase ration dialing to 3 then calls go some up but then again come back to lower value. For example I see vicidial in realtime report sections shows 250 calls ringing but only 80 or 100 calls are hitting to my billing/fail over server. This failover server running voipswitch and has lan connection with all telelphony servers.

I tried to monitor screen session of ASTVDauto and found this ...

2016-09-19 16:24:26|SERVER CALLS PER SECOND MAXIMUM SET TO: 3 |333.333333333333||
2016-09-19 16:24:26|LIVE AGENTS LOGGED IN: 15 ACTIVE CALLS: 29|
2016-09-19 16:24:26|WARID_O 192.168.172.83: agents: 15 (READY: 6) dial_level: 2 (6|9|0) 0|
2016-09-19 16:24:26|WARID_O 192.168.172.83: Calls to place: 1 (30 - 29 [0 + 29||]) 30 |
2016-09-19 16:24:26|CAMPAIGN DIFFERENTIAL: 183.3 81.1 (81.15 - 0.05)|
2016-09-19 16:24:26|LOCAL TRUNK SHORTAGE: 0|0 (30 - 80)|
2016-09-19 16:24:26|WARID_O 192.168.172.83: CALLING|
|UPDATE vicidial_hopper set status='QUEUE', user='VDAD_192.168.172.83' where campaign_id='WARID_O' and status='READY' order by priority desc,hopper_id LIMIT 1|
hopper rows updated to QUEUE: |1|
|SELECT lead_id,alt_dial FROM vicidial_hopper where campaign_id='WARID_O' and status='QUEUE' and user='VDAD_192.168.172.83' order by priority desc,hopper_id LIMIT 1|
|UPDATE vicidial_hopper set status='INCALL' where lead_id='17580730'|
hopper row updated to INCALL: |1|17580730|
2016-09-19 16:24:26|| number call dialed|WARID_O|V9191624260017580730|INSERT INTO vicidial_manager values('','','2016-09-19 16:24:26','NEW','N','192.168.172.83','','Originate','V9191624260017580730','Exten: 8368','Context: default','Channel: Local/03223798750@default','Priority: 1','Callerid: "V9191624260017580730" <04232534444>','Timeout: 45000','','','','VDACnote: WARID_O|17580730|92|03223798750|OUT|NONE|50')|5.00|NONE||
TIME DEBUG: 5.00|5.00|0| GMT: 11:24
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580167|03224016414|XFER||
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580166|03224016473|XFER||
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580457|03223881132|XFER||
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580487|03223865624|XFER||
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580498|03223849611|XFER||
2016-09-19 16:24:27|| dead call vac XFERd do nothing|17580625|03223826633|XFER||
2016-09-19 16:24:27|| lastcalldate UPDATED 1|'1001'||
2016-09-19 16:24:27|| logindate UPDATED 1|'WARID_O'||
TIME DEBUG: 5.00|5.00|0| GMT: 11:24


at the same I captured this on asterisk console

Milvik-t5:~ # asterisk -rx "core show channels" | grep 03
Local/03223794735@de 03223794735@default: Ring Dial(sip/vpst3/03223794735,,tT
Local/03223794735@de 03223794735@default: Down (None)
SIP/vpst3-000009f1 03223795579@trunkinb Down AppDial((Outgoing Line))
SIP/vpst3-000009f2 03223794930@trunkinb Down AppDial((Outgoing Line))
SIP/1035-00000860 8600062@default:1 Up MeetMe(8600062,F)
SIP/vpst3-000009f3 03223794741@trunkinb Down AppDial((Outgoing Line))
SIP/vpst3-000009f4 03223794735@trunkinb Down AppDial((Outgoing Line))
Local/03223794741@de 03223794741@default: Ring Dial(sip/vpst3/03223794741,,tT
SIP/vpst3-000009f8 03223790722@trunkinb Down AppDial((Outgoing Line))
Local/03223794741@de 03223794741@default: Down (None)
SIP/vpst3-000009f9 03223786930@trunkinb Down AppDial((Outgoing Line))
SIP/901038-0000085b 8600063@default:1 Up MeetMe(8600063,F)
SIP/1696-0000097b 03225646781@default: Up Dial(sip/vpst3/03225646781,,tT
SIP/901031-000007f6 8600058@default:1 Up MeetMe(8600058,F)
SIP/1748-00000403 8600066@default:1 Up MeetMe(8600066,F)
Local/03223786930@de 03223786930@default: Down (None)
Local/03223786930@de 03223786930@default: Ring Dial(sip/vpst3/03223786930,,tT
Local/03223794930@de 03223794930@default: Down (None)
Local/03223794930@de 03223794930@default: Ring Dial(sip/vpst3/03223794930,,tT
Local/03223795579@de 03223795579@default: Down (None)
Local/03223795579@de 03223795579@default: Ring Dial(sip/vpst3/03223795579,,tT
Local/03223790722@de 03223790722@default: Ring Dial(sip/vpst3/03223790722,,tT
Local/03223790722@de 03223790722@default: Down (None)
SIP/1976-00000903 8600060@default:1 Up MeetMe(8600060,F)

so ASTVDauto session shows 29 active calls but asterisk showing only 7 calls placed to sip trunk. Max trunk is set to 50 and max CPS is 5. Can you please give me some guidelines.

VERSION: 2.12-493a
BUILD: 150703-2105
© 2015 ViciDial Group

[muyousif]

sorry MAX CPS is 3.

[Vince-0]

Calls not connecting could be many things.
What are your real time carrier stats after a while?

Increase your trunk and CPS on Vici server settings.

[muyousif]

I have increased trunks to 100 on each server and CPS to 10. But still there are slow dialing. My carrier stats is given below



But at the same time on billing or carrier server it is just showing 120 calls from vicidial. But I can see vicidial showing almost 400 calls ringing so it is just confusing me. Even more than 100 agents are waiting I am getting very low number of calls on billing server. My agents wait time increasing even sometime upto 10 minutes. Thanks for help.

[njujihara]

What the vendor tells you?

[williamconley]

Change your calls per second to the max that your vendor will allow.

Check the asterisk cli for live calls "core show channels" (remember that agents in meetme rooms are also calls, so there'll be some math involved, LOL)

Change your drop call seconds to ZERO on the campaign. This has an unusual and undocumented effect of causing calls to regenerate more quickly and reliably.

After all this, you'll notice that the calls in the CLI more closely match the calls listed in the "calls being place" and "agents in calls" numbers.