vicidial.org

I've got the latest version running straight from the GoAutoDial website.

When I setup a campaign in manual dial mode everything works fine, but when I switch it to any type of auto dial mode, not a lot happens. The Asterix CLI shows that calls are being made, timing out, then dialling the next number. However, the problem is that the calls are not being passed to the agent screen or the agents phone.

Switching back to manual again and all works fine. Can anyone point me in the right direction? I'm assuming that the agent screen will update with the client info when Asterix is making the call in readiness for it to be answered?

Cheers

Post an example of this happening with the agiout logfile output for that example

Hi....

I have the same issue above except I am running VERSION: 2.4-309a BUILD: 110430-1642 ViciBox, it shows XX calls ringing but the agents do not receive the calls, only a limited amount of calls are getting into the agents and they are waiting 1-4 minutes between calls. When I stop the Campaign and all agents are logged out, it will still display the XX calls rining, almost as if the dialer is "stuck".

28 calls being placed 28 calls ringing 0 calls waiting for agents 0 calls in IVR NO AGENTS ON CALLS


Any ideas as to what would cause this ?
THANK YOU in advance for any assistance

output of screen -ls ?

Porpoise,

If you have access to the asterisk cli ("asterisk -r") take a look at the response you're getting back from the carrier when this is happening, also, if you have AMD (Answering Machine Detection) enabled you might have to up the campaigns dialing ratio as it will dump any call it identifies as an answering machine unless otherwise specified, since you are getting calls in both instances (manual and auto) you're basically down to carrier issues or need to allot more resources.

geoffers,

Sorry, not familiar with goautodial, however similar principles should apply given that when you're manually dialing the agents do receive calls (no detection enabled so the call will just ring through to the agent whatever the result) IF you aren't getting ANY calls in autodial, Autodialing should have parameters regarding what to do when you actually get a "live" call, these might not be set correctly (send to next available, longest wait, pre-recorded message etc). The agent screen will get cust info when the call is actually sent to that agent.

Thanks islandcabby.
In order to test this I've set every phone number in my campaign to my own DDI number. The Asterix real time logs show the calls being made and my phone rings. When I answer the call it times out after 5 seconds (this is the time period I've set to drop a call). When I answer the call as I would any call the system doesn't appear to recognise the fact that I'm talking and class it as a live call thus passing it to the agent screen as a live call. All these calls are then registered as a dropped call, and the next number is then dialled. I've set the agent to be passed the call based on longest wait.
I'm testing with 1 agent only currently.

Islandcabby,

THANK YOU for the heads up as it was a corrupt file which has since been repaired thanks to ALL the wonderful people here on the Forum.
I TOTTALLY appreciate everyone's assistance in getting this resolved.

Have a GREAT day/night and again THANKs

geoffers1 wrote:Thanks islandcabby.
In order to test this I've set every phone number in my campaign to my own DDI number. The Asterix real time logs show the calls being made and my phone rings. When I answer the call it times out after 5 seconds (this is the time period I've set to drop a call). When I answer the call as I would any call the system doesn't appear to recognise the fact that I'm talking and class it as a live call thus passing it to the agent screen as a live call. All these calls are then registered as a dropped call, and the next number is then dialled. I've set the agent to be passed the call based on longest wait.
I'm testing with 1 agent only currently.

Clarifying Questions:

1. Am I understanding correctly that you're dialing yourself?
2. Why do you have it set to drop the call after 5 seconds?
3. The system IS supposed to classify as live because you answered (in this case you are the "customer" or sales prospect), it then places the call in queue to the agent, however if it's set to time out it would of course end up being dropped and that would be the dispo the agent would use, can you take off the timeout and try again?

geoffers1

when you post, please post your entire configuration including (but not limited to) your installation method (which would be Goautodial, but you did not mention the Version of Goautodial) and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600
_____________

Yes, I say that to everybody. It helps as we delve into the issues.

What is your output from:
screen -list

Have you modified your system's IP address after the installation? (ie: is the "IP" listed for all phones,conferences,servers,ect in the Admin site the ip of your actual server?)

Hi. I've not changed the IP address of the server, all is as it installed.

Where do I run "screen -list" from?

If I run it from the Linux console I get "No Sockets found in /var/run/screen/S-apache"

If I run it from Asterix CLI I get "no such command"

Dial plan entry is:

exten => _944X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944X.,2,Dial(SIP/${EXTEN:3}@Node4,,tToR)
exten => _944X.,3,Hangup

Asterix output looks like this:

[May 9 11:36:22] WARNING[15261] file.c: Unexpected control subclass '-1'
[May 9 11:36:22] VERBOSE[15261] logger.c: [May 9 11:36:22] -- Executing [8368@default:2] AGI("Local/94401214211553@default-cd1d,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:22] VERBOSE[15261] logger.c: [May 9 11:36:22] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:22] VERBOSE[15261] logger.c: [May 9 11:36:22] -- Executing [8368@default:3] AGI("Local/94401214211553@default-cd1d,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[May 9 11:36:22] VERBOSE[15261] logger.c: [May 9 11:36:22] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[May 9 11:36:23] VERBOSE[15261] logger.c: [May 9 11:36:23] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[May 9 11:36:23] VERBOSE[15261] logger.c: [May 9 11:36:23] -- Executing [8368@default:4] AGI("Local/94401214211553@default-cd1d,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[May 9 11:36:23] VERBOSE[15261] logger.c: [May 9 11:36:23] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[May 9 11:36:24] VERBOSE[15206] logger.c: [May 9 11:36:24] == Manager 'sendcron' logged off from 127.0.0.1
[May 9 11:36:24] VERBOSE[15261] logger.c: [May 9 11:36:24] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[May 9 11:36:24] VERBOSE[15261] logger.c: [May 9 11:36:24] -- Executing [8368@default:5] Hangup("Local/94401214211553@default-cd1d,1", "") in new stack
[May 9 11:36:24] VERBOSE[15261] logger.c: [May 9 11:36:24] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/94401214211553@default-cd1d,1'
[May 9 11:36:24] VERBOSE[15261] logger.c: [May 9 11:36:24] -- Executing [h@default:1] DeadAGI("Local/94401214211553@default-cd1d,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 9 11:36:25] VERBOSE[15261] logger.c: [May 9 11:36:25] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[May 9 11:36:25] VERBOSE[15207] logger.c: [May 9 11:36:25] -- Executing [h@default:1] DeadAGI("Local/94401214211553@default-cd1d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----3") in new stack
[May 9 11:36:26] VERBOSE[15207] logger.c: [May 9 11:36:26] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----3 completed, returning 0
[May 9 11:36:26] VERBOSE[15207] logger.c: [May 9 11:36:26] == Spawn extension (default, 94401214211553, 2) exited non-zero on 'Local/94401214211553@default-cd1d,2'
[May 9 11:36:33] VERBOSE[15455] logger.c: [May 9 11:36:33] == Parsing '/etc/asterisk/manager.conf': [May 9 11:36:33] VERBOSE[15455] logger.c: [May 9 11:36:33] Found
[May 9 11:36:33] VERBOSE[15455] logger.c: [May 9 11:36:33] == Manager 'sendcron' logged on from 127.0.0.1
[May 9 11:36:33] VERBOSE[15456] logger.c: [May 9 11:36:33] -- Executing [94401213826567@default:1] AGI("Local/94401213826567@default-c11c,2", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:33] VERBOSE[15456] logger.c: [May 9 11:36:33] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:33] VERBOSE[15456] logger.c: [May 9 11:36:33] -- Executing [94401213826567@default:2] Dial("Local/94401213826567@default-c11c,2", "SIP/01213826567@Node4||tToR") in new stack
[May 9 11:36:33] VERBOSE[15456] logger.c: [May 9 11:36:33] -- Called 01213826567@Node4
[May 9 11:36:35] VERBOSE[15456] logger.c: [May 9 11:36:35] -- SIP/Node4-00000431 is making progress passing it to Local/94401213826567@default-c11c,2
[May 9 11:36:38] VERBOSE[15515] logger.c: [May 9 11:36:38] == Parsing '/etc/asterisk/manager.conf': [May 9 11:36:38] VERBOSE[15515] logger.c: [May 9 11:36:38] Found
[May 9 11:36:38] VERBOSE[15515] logger.c: [May 9 11:36:38] == Manager 'sendcron' logged on from 127.0.0.1
[May 9 11:36:38] VERBOSE[15516] logger.c: [May 9 11:36:38] -- Executing [94401213845796@default:1] AGI("Local/94401213845796@default-0f9c,2", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:38] VERBOSE[15516] logger.c: [May 9 11:36:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:38] VERBOSE[15516] logger.c: [May 9 11:36:38] -- Executing [94401213845796@default:2] Dial("Local/94401213845796@default-0f9c,2", "SIP/01213845796@Node4||tToR") in new stack
[May 9 11:36:38] VERBOSE[15516] logger.c: [May 9 11:36:38] -- Called 01213845796@Node4
[May 9 11:36:38] VERBOSE[15518] logger.c: [May 9 11:36:38] == Parsing '/etc/asterisk/manager.conf': [May 9 11:36:38] VERBOSE[15518] logger.c: [May 9 11:36:38] Found
[May 9 11:36:38] VERBOSE[15518] logger.c: [May 9 11:36:38] == Manager 'sendcron' logged on from 127.0.0.1
[May 9 11:36:38] VERBOSE[15519] logger.c: [May 9 11:36:38] -- Executing [94401214212475@default:1] AGI("Local/94401214212475@default-ffa2,2", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:38] VERBOSE[15519] logger.c: [May 9 11:36:38] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:38] VERBOSE[15519] logger.c: [May 9 11:36:38] -- Executing [94401214212475@default:2] Dial("Local/94401214212475@default-ffa2,2", "SIP/01214212475@Node4||tToR") in new stack
[May 9 11:36:38] VERBOSE[15519] logger.c: [May 9 11:36:38] -- Called 01214212475@Node4
[May 9 11:36:40] VERBOSE[15519] logger.c: [May 9 11:36:40] -- SIP/Node4-00000433 is making progress passing it to Local/94401214212475@default-ffa2,2
[May 9 11:36:40] VERBOSE[15516] logger.c: [May 9 11:36:40] -- SIP/Node4-00000432 is making progress passing it to Local/94401213845796@default-0f9c,2
[May 9 11:36:52] VERBOSE[15456] logger.c: [May 9 11:36:52] -- SIP/Node4-00000431 answered Local/94401213826567@default-c11c,2
[May 9 11:36:52] VERBOSE[15455] logger.c: [May 9 11:36:52] > Channel Local/94401213826567@default-c11c,1 was answered.
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- Executing [8368@default:1] Playback("Local/94401213826567@default-c11c,1", "sip-silence") in new stack
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- Playing 'sip-silence' (language 'en')
[May 9 11:36:52] WARNING[15670] file.c: Unexpected control subclass '-1'
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- Executing [8368@default:2] AGI("Local/94401213826567@default-c11c,1", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- Executing [8368@default:3] AGI("Local/94401213826567@default-c11c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[May 9 11:36:52] VERBOSE[15670] logger.c: [May 9 11:36:52] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[May 9 11:36:53] VERBOSE[15685] logger.c: [May 9 11:36:53] == Parsing '/etc/asterisk/manager.conf': [May 9 11:36:53] VERBOSE[15685] logger.c: [May 9 11:36:53] Found
[May 9 11:36:53] VERBOSE[15685] logger.c: [May 9 11:36:53] == Manager 'sendcron' logged on from 127.0.0.1
[May 9 11:36:53] VERBOSE[15686] logger.c: [May 9 11:36:53] -- Executing [94401214224680@default:1] AGI("Local/94401214224680@default-5e0d,2", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:53] VERBOSE[15686] logger.c: [May 9 11:36:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:53] VERBOSE[15686] logger.c: [May 9 11:36:53] -- Executing [94401214224680@default:2] Dial("Local/94401214224680@default-5e0d,2", "SIP/01214224680@Node4||tToR") in new stack
[May 9 11:36:53] VERBOSE[15686] logger.c: [May 9 11:36:53] -- Called 01214224680@Node4
[May 9 11:36:53] VERBOSE[15688] logger.c: [May 9 11:36:53] == Parsing '/etc/asterisk/manager.conf': [May 9 11:36:53] VERBOSE[15688] logger.c: [May 9 11:36:53] Found
[May 9 11:36:53] VERBOSE[15688] logger.c: [May 9 11:36:53] == Manager 'sendcron' logged on from 127.0.0.1
[May 9 11:36:53] VERBOSE[15689] logger.c: [May 9 11:36:53] -- Executing [94401213826763@default:1] AGI("Local/94401213826763@default-8462,2", "agi://127.0.0.1:4577/call_log") in new stack
[May 9 11:36:53] VERBOSE[15689] logger.c: [May 9 11:36:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 9 11:36:53] VERBOSE[15689] logger.c: [May 9 11:36:53] -- Executing [94401213826763@default:2] Dial("Local/94401213826763@default-8462,2", "SIP/01213826763@Node4||tToR") in new stack
[May 9 11:36:53] VERBOSE[15689] logger.c: [May 9 11:36:53] -- Called 01213826763@Node4
[May 9 11:36:53] VERBOSE[15670] logger.c: [May 9 11:36:53] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[May 9 11:36:53] VERBOSE[15670] logger.c: [May 9 11:36:53] -- Executing [8368@default:4] AGI("Local/94401213826567@default-c11c,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[May 9 11:36:53] VERBOSE[15670] logger.c: [May 9 11:36:53] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[

under root

Code: Select all

screen -ls

I get "No Sockets Found in var/run/screen/S-apache"

that BIG is a problem

kindly reboot your server then check again

Ok, but it'll have to wait until the agents are on a break. Will let you know when it's done.

geoffers1 wrote:I get "No Sockets Found in var/run/screen/S-apache"

Are you logged in as apache!?

Ooops, my bad.

Logged in as root and here is the screen -ls dump:

[root@go ~]# screen -ls
There are screens on:
3589.ASTVDadapt (Detached)
3368.asterisk (Detached)
3577.ASTsend (Detached)
3580.ASTlisten (Detached)
3583.ASTVDauto (Detached)
3362.astshell20120510084630 (Detached)
3574.ASTupdate (Detached)
3599.goautodial_d (Detached)
3592.ASTfastlog (Detached)
3586.ASTVDremote (Detached)
10 Sockets in /var/run/screen/S-root.

[root@go ~]#

Hi! I am completely new to this GO AUTO DIAL setup and all.

I some how found that the calls are ringing but the calls are not pasing to the agents.
Then checked "aestricks -r"
that gave me WARNING(13206) unable to right to channel

i was previously using the new version of go auto dial (3.0) in that everything was fine so i used the same sip registration settings in the version 2.1 .
So this sip should work
Now please any 1 help pe in detailed command how can i fix this warning.
So that i may get calls to my agents screen

And when i tried manually it gave me error code 20

And also it says RTP 126

Hope u al MASTER MINDS got my issue. Please gyus help me out to learn this.