Cannot Use Listen-Monitor Feature GAD or ViciDial

[escondido]

Long time lurker, first time poster.

It's my understanding that I can BARGE in on any active call that I want having admin access. However, LISTEN allows me to continually monitor what the agent is hearing on their side, but they cannot hear me. So, in theory, I could LISTEN for hours if I wanted and hear the conversations, but BARGE only works on a per call basis. If wrong, please correct me.

I've tried to Listen/Monitor or Barge using 2 different Admin accounts that work fine when logging into the GAD Agent Login.

In GAD v3.3 Clicking Agent Monitor> Agent Name> Barge> Enter Admin extension (tried 2 different admin phones works well. I see, "VC Blind Monitor" populate in Zoiper, click accept. This feature also works fine, via the Real-Time Main Report. Also works dialing the SessionID with Zoiper.

However, selecting LISTEN in the same section for GAD doesn't do anything, nor does selecting MONITOR in the ViciDial report. And finally, adding a 0 as a prefix to the SessionID also doesn't do anything. I've been expecting Zoiper to ring, but nothing is happening. It's very weird.

Does anybody know what might be causing the issue and how to fix it?

Thanks.

[williamconley]

1) Good job posting your specs.

2) Goautodial is not great with multi-server. I would have to ask what instructions you used to configure a multi-server GAD system and confirm that the attempts you made were for calls on the same server (ie: Agent and Admin and Prospect all on the same dialer, with no legs of any of the calls on another server).

3)

And finally, adding a 0 as a prefix to the SessionID also doesn't do anything. I've been expecting Zoiper to ring, but nothing is happening. It's very weird

This is a little contradictory: If you dialed "0" and then the session ID in Zoiper, why would zoiper ring? Zoiper is making a call, not taking a call in this scenario.) So the question then is how you were dialing, or where you put this information?

4) None of what you described is related to the Vicidial method for monitor/barge/whisper (except the direct dial from Zoiper). Please locate the stock Vicidial screens (they are on the server, you just have to find them, GAD does also install the Stock Vicidial system in addition to the GAD modified pages!) and try to use the Vicidial Manager's Manual method for monitor/barge/whisper which applies to your Version of Vicidial.

Hint: It's in the Real Time screen of Vicidial and involves choosing the report options and putting in your phone extension ...

Happy Hunting!

[escondido]

I appreciate the feedback William.

I might not have been clear about what I did in my original post. I'm using GAD strictly for outbound calls. I haven't messed with the Inbound settings yet. I know that I should have 2 methods to monitor/barge. Either using Real-Time Reports or dialing directly from Zoiper.

For argument sake, let's assume my Phone Extension is 55001 and it's setup as a SIP account since I use it for dialing myself. It's also a backup admin account where 1 is selected for every option I could find. I'm trying to monitor/barge an agent of mine with a SessionID of 8600051

I just ran a new test after confirming 55001 was properly registered with Zoiper.

Choose Real-Time Report and then selected "Choose Report Display Options"
Screen Refresh= 4 seconds
Inbound=Yes (don't really care about this since the system is strictly for outbound calling)
Monitor= BARGE
Phone= 55001

This producers the Listen and Barge links next to my agent. I click Barge and Zoiper rings, I accept the call and can hear my agent talking with a client. Using the same Zoiper Sip account I manually dial 8600051. That also works perfectly when the agent is on a live call. It feels like it's a similar process when logging into the Agent Login and having to accept the call before hearing the lovely, "you're the only person in this conference."

However, clicking Listen still also shows the success pop up box, but nothing happens at all. So I try to manually dial 08600051 from the same sip account using Zoiper and I see the following message, "Sip 603 - Declined Call Rejected." I have no idea why that is happening because Barge works fine.

Lastly, I tried to dial 8162 from Zoiper and saw the exact same error message listed below.

Maybe it's a carrier issue or bug. I've been making outbound calls or 3 weeks using the same settings, so I'm scratching my head.

[williamconley]

Maybe it's a carrier issue

Your carrier is not involved in this call at all.

Code: Select all

grep Meetme /etc/asterisk/extensions.* -Rn

This will expose the extensions available for Meetme entry. 76800051, 68600051, 58600051, etc. Each of those lines should have an explanation for its purpose directly above it. For instance:

; quiet monitor-only extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Meetme(${EXTEN:1},Fmq)
exten => _68600XXX,n,Hangup()

[escondido]

This what mine shows

/etc/asterisk/extensions.ael:444:// 8600 => Meetme(1234);
/etc/asterisk/extensions.conf:352:exten => _86000[0-4]X,1,Meetme(${EXTEN},q)
/etc/asterisk/extensions.conf:356:exten => _86000[5-9]X,1,Meetme(${EXTEN},F)
/etc/asterisk/extensions.conf:358:exten => _8600[1-2]XX,1,Meetme(${EXTEN},F)
/etc/asterisk/extensions.conf:362:exten => _78600XXX,1,Meetme(${EXTEN:1},Fq)
/etc/asterisk/extensions.conf:366:exten => _68600XXX,1,Meetme(${EXTEN:1},Fmq)
/etc/asterisk/extensions.conf:370:exten => _58600XXX,1,Meetme(${EXTEN:1},Fmq)
/etc/asterisk/extensions.conf:374:exten => _558600XXX,1,Meetme(${EXTEN:2},Fmq)
/etc/asterisk/extensions.conf:378:exten => _86009XX,1,Meetme(${EXTEN},Fmq)
/etc/asterisk/extensions.conf:380:exten => _986009XX,1,Meetme(${EXTEN:1})
/etc/asterisk/extensions.conf:650:exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},mqX) ; Listen
/etc/asterisk/extensions.conf:652:exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},X) ; Barge

I'm seeing a busy/congestion error message when dialing 8162. Which is very odd.

[williamconley]

Perhaps the extension you are dialing from is not in the "default" context. Show CLI output from a single attempt (not 3000 lines of unrelated code, but ...) don't leave out any parts of the attempted call: start to finish.

Have you edited your extensions.conf file? Are there any errors during asterisk loading? (screen -r asterisk, to be sure you catch ALL errors, even perl script errors)

[escondido]

The only issue with this test server is watching the "lovely" folks trying to register SIP's and/or the Web GUI using a brute force attack. The latter automatically gets redirected to a variety of "creative" websites after a failed login attempt

I am a seriously done messing around it. Moving over to Vicibox.

I appreciate the help though.

[thephaseusa]

I see you are setting up vicibox 8.1.2 . Be sure and get the whitelist/dynamic firewall set up. And the voip thieves won't be able to run brute force password attacks against your sip.

JM