Timeout issue on 'Manual Dial'

[nobesnickr]

I'll start out by thanking everyone so far for all there help to get me caught up around here.

My next kind of annoying thing is the double standard I seem to be coming across when we are on a predictive dialing campaign. We have the call timeout setting to 25 seconds to ward of answering machines but when my reps try to do manual calls to call someone back they run into the same timeout issues.

Is this something the system should be separating automatically? If not, is there away to not have them abide by the same standards?

Thanks again!

[mflorell]

What is your Dial timeout set to in extensions.conf?

What is your dial_timeout set to in your campaign?

[nobesnickr]

The timeout is set to 25 seconds on the campaign but not set in the extensions.conf (as far as I can tell)

[mflorell]

When a Manual dial call is placed, what is the output of the Dial command on the Asterisk CLI?

[nobesnickr]

probably a lot more than you were wanting but I'd rather give you a little more then not enough... Thank again!


== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
[Nov 28 11:12:30] NOTICE[25775]: channel.c:3051 __ast_request_and_dial: Unable to request channel SIP/pmccaffrey
[Nov 28 11:12:30] WARNING[25775]: cdr.c:830 ast_cdr_init: CDR already initialized on '**Unknown**'
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600054@default:1] MeetMe("Local/8600054@default-25e0,2", "8600054") in new stack
> Channel Local/8600054@default-25e0,1 was answered.
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1013 for conference '8600054'
-- Executing [94802097440@default:1] AGI("Local/8600054@default-25e0,1", "agi://127.0.0.1:4577/call_log") in new stack
-- <Local/8600054@default-25e0,2> Playing 'conf-onlyperson' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [94802097440@default:2] Dial("Local/8600054@default-25e0,1", "Zap/g1/4802097440") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/4802097440
-- Zap/1-1 is proceeding passing it to Local/8600054@default-25e0,1
[Nov 28 11:12:39] WARNING[25786]: file.c:1078 waitstream_core: Unexpected control subclass '15'
-- Zap/1-1 is making progress passing it to Local/8600054@default-25e0,1
[Nov 28 11:12:39] WARNING[25786]: file.c:1078 waitstream_core: Unexpected control subclass '14'
-- Zap/1-1 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [78600054@default:1] MeetMe("Local/78600054@default-8711,2", "8600054|q") in new stack
> Channel Local/78600054@default-8711,1 was answered.
-- Executing [8309@default:1] Answer("Local/78600054@default-8711,1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/78600054@default-8711,1", "wav|7001_4802097440_20071128-111247") in new stack
-- Executing [8309@default:3] Wait("Local/78600054@default-8711,1", "3600") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Zap/1-1 answered Local/8600054@default-25e0,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[mflorell]

Are you using Analog lines or T1/E1s?

[nobesnickr]

4 PRI T1's

[mflorell]

I'm not seeing a Zap/1-1 hangup in your output, where does it get hung up?

[nobesnickr]

That is part of the problem. After 50 seconds Vicidial will pop-up saying "Dialed Timed Out, Contact your administrator" but keeps the call open and seems to put the user back into the queue to take new calls and then overlays calls into the meetme room.

I have a feeling some of this might be cause from errors during install (and the fact that I am using 1.4.x) so I will be redoing the server this weekend but I'd like to avoid this in the future lol.



On a second thought, I wonder if this issue is related to another glitch I seem to be having where the server load skyrockets to 8-14+ even though we have only 4-6 agents on at any given time and calls start having issues hanging up and overlapping with new calls. A reset of asterisk fixes the problem right away but it does get kind of annoying. I am starting to get the feeling these issues maybe connected.

Like I said, I am going to reinstall this weekened with Asterisk 1.2.x and Mysql 5.x and also use Fedora 7 instead of Ubuntu and see how all that helps.

I really do, as always, appreciate your help on this.

[nobesnickr]

I just noticed (and this probably has something to do with the issues) that on outbound calls the "LIVE CALL" light doesnt turn green, it just always says NO LIVE CALL

I am fairly certain these are connected somehow but reading through the forums hasnt turned up any leads.

Thank you again for all your help

[osib]

After all these years ViciDial still have issues with "Dialed Timed Out, Contact your administrator" which comes in Manual Dial after 50 secs and the screen shows waiting for ringing... Is there anyway to fix this?

Thanks

[mflorell]

Yep, we still haven't found a way to force customers to pick up the phone

How would you fix this?

[osib]

Hi mflorell

We can't force customer to pick up the call. Wish we could

Anyways, the problem is After 50 secs, it comes with the popup "Dialed Timed Out, Contact your administrator" in agent's screen but the phone is still connected and ringing and if customers picks up after 50 secs then the agent can talk with customer.

The solution I was looking for is something like be able to increase or decrease the 50sec limit and then after the limit hangup the call completely and show the message "Dialed Timed Out" and take the action to dispo screen for agent, as administrator cannot force customer to pick up as well

Thanks

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method 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 your installation is "Hosted" list the site name of the host.

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

3) If this works as you describe, it would require programming the agent screen to be aware of the dial timeout setting in your dialplan. Note that the auto-dial "dial timeout" is not related because it's not autodialed. So the manual dial's dial timeout would be required ... but that's not a setting in Vicidial, it's mixed into the Dialplan settings of the carrier, this field is not interpreted at all.

4) Of course, you could set the dial timeout for manual dialed calls to be 50 seconds and then it would coincide with this warning and the call would terminate at the same time. This would be set by setting the manual dial prefix to a different carrier and specifying the dial timeout in the carrier setting's "dial" command.

5) Alternately, the javascript on the agent page would need to be altered to a different hard-coded number OR somehow be "aware" of the dial timeout for this carrier.

6) Or you could pay someone to make the "dial timeout" setting for manually dialed calls conform to the auto-dialed calls dial timeout (or try this yourself somewhere in the AJAX)

7) Or you could pay someone to add a new preference for manual-dial calls to have their own dial timeout per campaign (or try this yourself somewhere in the AJAX).

[osib]

ASTGUICLIENT 2.11rc1
Asterisk 1.8.23.0-vici
debian 8.2.0
multiserver with separate web, db, telephony in XenServer 6.5.0

[williamconley]

ASTGUICLIENT 2.11rc1
Asterisk 1.8.23.0-vici
debian 8.2.0
multiserver with separate web, db, telephony in XenServer 6.5.0

Vicidial Version with build (rquired).

Link to instructions used to install, since you obviously did not use the Vicibox .iso or Goautodial .iso. Could be quite useful to include in your signature one day. Just sayin'

[osib]

ViciDial VERSION: 2.11-476a
BUILD: 150302-0945

[mflorell]

Have you tried raising the dial timeout in your campaign settings?

[osib]

Hi mflorell
I have tried raising the Dial Timeout in campaign settings. If I raise it to say 60sec then while manual dialling "Dialed Timed Out, Contact your administrator" comes after 60 sec. But this is not the solution I am after as the campaign on use is with Dial Method: RATIO

I also have tried with DialPlan Entry in Carriers in Dial(). This hangsup the call in asterisk but Vicidial... in Agent screen doesn't recognise that the call has already been hung up and still shows Waiting for ringing... Asterisk shows "...making progress passing it..." but doesn't go to ringing though can hear ringing tone.

I was wondering if there are any tweaks so while manual dialling rather than showing "Dialed Timed Out, Contact your administrator" disconnect the call take it to dispo screen.

Thanks

[williamconley]

Pardon the continued intrusion, but this appears to mean that you would want a new preference for manual dial call timeout (and then modify the agent javascript to grab this value plus using it when dialing).

Since the agent screen is already "checking a value", this is not nearly as intense as having to code the ability to get that value in the first place.

I'd suggest an upgrade, but I do not believe this feature has been modified recently. I could be wrong about that, of course, as I have not checked!

[osib]

Hi williamconley
Suggesstions are always useful. I think astguiclient_2.11rc1 is the latest version.

Thanks

[williamconley]

Not likely. I'm on a machine editing files with:

Code: Select all

2.12-505a build 150806-1348

[mflorell]

That would be a new feature, please post that as a feature request to the Issue Tracker and post your patch files for the change, or consider sponsoring development.

[osib]

Hi mflorell
First of all thanks for the new feature of Manual Dial Time out but the problem is the time defined is not functioning. I can hear the ringtone but vicidial shows Waiting for Ringing and still the call is connect after the defined manual dial timeout till 50 sec and then "Dialed Timed Out, Contact your administrator" pops up. I have Dial timeout in campaign setting as 26 and Manual Dial Timeout = 35

Please advise

Thank you

[williamconley]

... I have Dial timeout in campaign setting as 26 and Manual Dial Timeout = 35 ...

Manual Dial Timeout? Is that a setting now? (I Haven't Looked ... sorry)

If you want to force dial timeout, you can set it in the Carrier directly. But you'll want to use a separate carrier just for manual dials in that case to avoid breaking Campaign Dial Timeout. Note that the second carrier can still be through the same telephone company, so there wouldn't need to be anything else different about the call. Just modify the "dial" command in the Dialplan Extension for a fresh carrier to include the dial timeout (google it, "dial" is a standard asterisk command! Its use is fairly easy.)