Drop vs disconnected

[bc319]

VERSION: 2.6-365a/BUILD: 120420-1620

I was unable to find this topic so I apologize if this has been addressed in another thread.

We are experiencing a lot of "dropped" calls that are, in fact, disconnected numbers. Is there a way to pass the actual call so the agent can hear the "The number you are calling is not in service" message instead of the simulated ring and drop that the dialer uses? From a support standpoint, I'm just looking for a way to minimize false alarms.

Thanks in advance.

[omarrodriguezt]

Notice that dropped calls are calls that never were answered by the agents. Increment the dial time out (go to campaign detail). Also change the routing extension:
Routing Extension - This field allows for a custom outbound routing extension. This allows you to use different call handling methods depending upon how you want to route calls through your outbound campaign. Formerly called Campaign VDAD extension.
- 8364 - same as 8368
- 8365 - Will send the call only to an agent on the same server as the call is placed on
- 8366 - Used for press-1, broadcast and survey campaigns
- 8367 - Will try to first send the call to an agent on the local server, then it will look on other servers
- 8368 - DEFAULT - Will send the call to the next available agent no matter what server they are on
- 8369 - Used for Answering Machine Detection after that, same behavior as 8368
- 8373 - Used for Answering Machine Detection after that same behavior as 8366
- 8374 - Used for press-1, broadcast and survey campaigns with Cepstral Text-to-speech
- 8375 - Used for Answering Machine Detection then press-1, broadcast and survey campaigns with Cepstral Text-to-speech

[williamconley]

VERSION: 2.6-365a/BUILD: 120420-1620

I was unable to find this topic so I apologize if this has been addressed in another thread.

We are experiencing a lot of "dropped" calls that are, in fact, disconnected numbers. Is there a way to pass the actual call so the agent can hear the "The number you are calling is not in service" message instead of the simulated ring and drop that the dialer uses? From a support standpoint, I'm just looking for a way to minimize false alarms.

Thanks in advance.

Your carrier is passing the calls as answered when they are disconnected? LOL Change carriers. Check the asterisk CLI in sip debug mode and see if they are marking them as "answered" and complain. Disconnected numbers cannot, by definition, answer.
___

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[mike.salazar]

Hi,

We are using survey method and we have a requirements that all disconnected numbers with a message saying "The number you have dialed is not in service" should be sent to the extension instead of being tag as "NA". Is there a way or settings in the vicidial to accomplish this? Thanks in advance.

[lark]

under Admin > System Settings Look for > Enhanced Disconnect Logging cheers

[mike.salazar]

I have already done that but it did not help. when i dialed the number manually in vicidial it just keeps on ringing. but when i dialed the same number using freepbx with same version of asterisk i can hear the recording that is saying "The number you have dialed is not in service"

[lark]

you maybe force to have a Ring or your Provider force a Ring to generate

[mike.salazar]

what do you mean by "you maybe force to have a Ring"? I'm using same provider on my vicidial server and freepbx but when i dialed the same number i get different result. vicidial - ring only. freepbx plays the message that the number has been disconnected which is we need to capture in vicidial.

[williamconley]

Your challenge is that these calls are never technically "answered" and as such never sent to the "handle this call, it has been answered" script in Vicidial.

The way Vicidial works is that when calls are answered, the campaigns' "Routing Exten" takes over and the script associated with that routing exten "handles" the call. Since these calls are CANCEL or CONGESTION instead of ANSWER, they are never passed to this script for processing.

So, No, Vicidial is not able to manage these calls without customization.

[mike.salazar]

Since the Freepbx answers the channel. what i did is i created a trunk in my dialer that sends out all calls to the freepbx then from freepbx to the carrier. the call is now being answered in vicidial. when i do a manual dial i can now hear the disconnection recording. but when i do a auto dial it does not send the call to the extension even though the 8366 script was being called already by the dialer.

[williamconley]

Does it send calls that are answered by humans that do not press a button to the extension? If not, modify your setting for the survey.

[mike.salazar]

Yes, my settings are

Survey No-Response Action: OPTOUT
Survey Method: EXTENSION
Survey Survey Xfer Extension: agents extension

[williamconley]

In which case there must be a difference (detected by Vicidial) between that call and the "This number is disconnected" call and it is being treated differently. asterisk CLI output would likely show you the difference. Perhaps it is not as "answered" as you think it is ... but that is easily tested

[mike.salazar]

Here is the log when the dialer dials it using the freepbx as trunk

[Oct 25 08:12:31] -- Called 2032541100@freepbx
[Oct 25 08:12:31] -- SIP/freepbx-0000005a answered Local/912032541100@default-09a3,2
[Oct 25 08:12:31] > Channel Local/912032541100@default-09a3,1 was answered.
[Oct 25 08:12:31] -- Executing [8366@default:1] Playback("Local/912032541100@default-09a3,1", "sip-silence") in new stack
[Oct 25 08:12:31] -- <Local/912032541100@default-09a3,1> Playing 'sip-silence' (language 'en')
[Oct 25 08:12:31] WARNING[3840]: file.c:1292 waitstream_core: Unexpected control subclass '-1'
[Oct 25 08:12:31] -- Executing [8366@default:2] AGI("Local/912032541100@default-09a3,1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 25 08:12:31] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 25 08:12:31] -- Executing [8366@default:3] AGI("Local/912032541100@default-09a3,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Oct 25 08:12:31] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 25 08:12:32] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 25 08:12:32] -- Executing [8366@default:4] AGI("Local/912032541100@default-09a3,1", "agi-VDAD_ALL_outbound.agi|SURVEYCAMP-----LB") in new stack
[Oct 25 08:12:32] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Oct 25 08:12:33] -- Executing [h@default:1] DeadAGI("Local/912032541100@default-09a3,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----2") in new stack
[Oct 25 08:12:33] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 25 08:12:33] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---2-----2 completed, returning 0
[Oct 25 08:12:33] == Spawn extension (default, 912032541100, 2) exited non-zero on 'Local/912032541100@default-09a3,2'
[Oct 25 08:12:33] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Oct 25 08:12:33] -- Executing [8366@default:5] Hangup("SIP/freepbx-0000005a", "") in new stack
[Oct 25 08:12:33] == Spawn extension (default, 8366, 5) exited non-zero on 'SIP/freepbx-0000005a'
[Oct 25 08:12:33] -- Executing [h@default:1] DeadAGI("SIP/freepbx-0000005a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack


And here is the output when dialing directly from the carrier which is also the carrier that is being used by the freepbx.

[Oct 25 09:06:50] -- Executing [912032541100@default:1] AGI("SIP/6453-000000db", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 25 09:06:50] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 25 09:06:50] -- Executing [912032541100@default:2] Dial("SIP/6453-000000db", "SIP/2032541100@lataone||tTor") in new stack
[Oct 25 09:06:50] -- Called 2032541100@lataone
[Oct 25 09:06:51] -- SIP/lataone-000000dc is making progress passing it to SIP/6453-000000db
[Oct 25 09:07:22] -- Got SIP response 503 "Service Unavailable" back from 216.89.76.6
[Oct 25 09:07:22] -- SIP/lataone-000000dc is circuit-busy
[Oct 25 09:07:22] == Everyone is busy/congested at this time (1:0/1/0)
[Oct 25 09:07:22] -- Executing [912032541100@default:4] Hangup("SIP/6453-000000db", "") in new stack
[Oct 25 09:07:22] == Spawn extension (default, 912032541100, 4) exited non-zero on 'SIP/6453-000000db'
[Oct 25 09:07:22] -- Executing [h@default:1] DeadAGI("SIP/6453-000000db", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
[Oct 25 09:07:22] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[williamconley]

Since the Freepbx answers the channel.

This shows an answer, missing in the "failed" call:

Code: Select all

[Oct 25 08:12:31] > Channel Local/912032541100@default-09a3,1 was answered.

Based on the cli output, your assertion that "FreePBX answers the channel" is false. There is no answer. There is "CONGESTION", which is decidedly NOT answer. It is theoretically possible to continue the call in the Carrier Dialplan Entry even though it is "congested", but this may be a moot point as the call may have already been terminated (from the remote end). Or you can add "answer" to the FreePBX dialplan before it passes the call to the actual carrier.

[mike.salazar]

Hi William,

Apologize for the late response. Here is my dial plan in the freepbx.

[Dialer-140]
exten => _.,1,Answer()
exten => _.,n,NOop(${EXTEN})
exten => _.,n,Dial(SIP/CARRIER/${EXTEN})
;exten => s,n,Hangup()

[williamconley]

Cool. Now show CLI from the FreePBX box to demonstrate that FreePBX believes it answered the call. Based on the Vicidial CLI, this does not happen. So ... your dialplan entry in FreePBX may not be executing as you think it is. That's where my money is.