NO LIVE CALL & HANGUP CUSTOMER disabled

[spinto]

VERSION: 2.4-303a
BUILD: 110224-1427

1.4.39.1-vici

I don't believe the part in red below is supposed to happen when logging onto VICIdial. I'm able to manual dial, connect, and hear audio both ways except the HANGUP CUSTOMER button is disabled and it says NO LIVE CALL. I'm using an external phone.

Logon to VICIdial
[Apr 7 21:20:30] == Parsing '/etc/asterisk/manager.conf': [Apr 7 21:20:30] Found
[Apr 7 21:20:30] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 7 21:20:30] -- Executing [817863389999@default:1] AGI("Local/817863389999@default-44de,2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 7 21:20:30] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 7 21:20:30] -- Executing [817863389999@default:2] Dial("Local/817863389999@default-44de,2", "SIP/17863389999@nexvortex.com||tTo") in new stack
[Apr 7 21:20:30] -- Called 17863389999@nexvortex.com
[Apr 7 21:20:33] -- SIP/nexvortex.com-00000009 is making progress passing it to Local/817863389999@default-44de,2
[Apr 7 21:20:36] -- SIP/nexvortex.com-00000009 is making progress passing it to Local/817863389999@default-44de,2
[Apr 7 21:20:40] -- SIP/nexvortex.com-00000009 answered Local/817863389999@default-44de,2
[Apr 7 21:20:40] > Channel Local/817863389999@default-44de,1 was answered.
[Apr 7 21:20:40] -- Executing [8600051@default:1] MeetMe("Local/817863389999@default-44de,1", "8600051|F") in new stack
[Apr 7 21:20:40] == Parsing '/etc/asterisk/meetme.conf': [Apr 7 21:20:40] Found
[Apr 7 21:20:40] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 7 21:20:40] Found
[Apr 7 21:20:40] -- Created MeetMe conference 1023 for conference '8600051'
[Apr 7 21:20:40] -- <Local/817863389999@default-44de,1> Playing 'conf-onlyperson' (language 'en')
[Apr 7 21:20:40] WARNING[3349]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Apr 7 21:20:40] -- Executing [h@default:1] DeadAGI("Local/817863389999@default-44de,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----0") in new stack
[Apr 7 21:20:40] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----0 completed, returning 0
[Apr 7 21:20:40] == Spawn extension (default, 817863389999, 2) exited non-zero on 'Local/817863389999@default-44de,2'
[Apr 7 21:20:42] == Manager 'sendcron' logged off from 127.0.0.1

Make the call
[Apr 7 21:26:23] == Parsing '/etc/asterisk/manager.conf': [Apr 7 21:26:23] Found
[Apr 7 21:26:23] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 7 21:26:23] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-b1a7,2", "8600051|F") in new stack
[Apr 7 21:26:23] > Channel Local/8600051@default-b1a7,1 was answered.
[Apr 7 21:26:23] -- Executing [813054039999@default:1] AGI("Local/8600051@default-b1a7,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 7 21:26:23] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 7 21:26:23] -- Executing [813054039999@default:2] Dial("Local/8600051@default-b1a7,1", "SIP/13054039999@nexvortex.com||tTo") in new stack
[Apr 7 21:26:23] -- Called 13054039999@nexvortex.com
[Apr 7 21:26:24] -- SIP/nexvortex.com-0000000a answered Local/8600051@default-b1a7,1
[Apr 7 21:26:25] == Manager 'sendcron' logged off from 127.0.0.1

Misc
Load Avg is very low

Screen -r

22773.astshell20110406181238 (Detached)
2914.ASTfastlog (Detached)
2908.ASTVDauto (Detached)
2904.ASTsend (Detached)
2910.ASTVDremote (Detached)
2901.ASTupdate (Detached)
2783.asterisk (Detached)
2906.ASTlisten (Detached)
8294.ASTVDadapt (Detached)

The server was setup by someone else using VICIBox.

[boybawang]

does the phone you are on rings when you logon?
what browser are you using?
try using firefox

[spinto]

It rings. I also tried a couple different browsers and it still isn't working. Here's the trunk information:

Account Entry
[nexvortex.com]
username=auth_name
user=phone
type=peer
secret=sssssss
nat=yes
insecure=very
host=xx.xx.xx.xx
authname=auth_name
fromuser=auth_name
from_user=2222222222
callerid=company <2222222222>
dtmfmode=rfc2833
dtmf=inband
allow=all
context=trunkinbound
canreinvite=yes
qualify=yes

Dialplan Entry
exten => _81NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@nexvortex.com,,tTo)
exten => _81NXXXXXXXXX,3,Hangup

[boybawang]

try reinstalling the astguiclient scripts

[spinto]

No luck. Its still happening.

[williamconley]

1) Please always post your installation method

2) It appears you are trying to log in to your system with an extenal phone? (ie: Local/817863389999@default-44de,1> Playing 'conf-onlyperson)? Why is it playing the only person sound to the phone number if YOU aren't that phone number? Did you hang up the sip phone or is there no sip phone involved?

3) You did not use a global variable, this leads to bad habits and unnecessary hair-pulling. you should change to the standard method until after you have your system online solidly and want to begin designing your own wheel. Makes life simpler. TRUNK8=SIP/nexvortex and then make the context nexvortex instead of nexvortex.com. Then the system will associate things properly, and you can put the standard dial command in Dial(${TRUNK8}/${EXTEN:1},,To)

Then clarify for me (please ): You log in as an agent. You answer. You hear "You are the only person in this conference" and THEN you press Resume (without hanging up the phone). right?

[spinto]

1.) The system was not installed by us. They installed it using vicibox and we were retained to get it up and running.

2.) We are trying to logon using an external phone. The reason is they are behind a firewall and I cannot hook up a remote sip phone to the system. There is currently no SIP phone involved.

3.) The sip setting and extensions settings were added using the VICIdial carrier section in the vicidial admin panel. I don't see how using a global variable will fix anything. We use those settings across different system succesfully.

To clarify, you are correct about the logon method, except since its a manual campaign, there's no resume dial. Also, I use the manual link (at the bottom of the vicidial screen) to test it out. It dials and connects fine. I can hear both sides of the conversation, but the hangup button is disabled.

[williamconley]

vicibox is a good start. when you log in with ssh, the splash screen should tell you which version it is.

If you do not have a local sip phone to test, use an external phone set up under admin->phones. this will still allow the standard "log in and hear you are the only person" method. Remember that to VICIDIAL any phone in this list is like any other (your cell, IAX, SIP, anything that can be put into the admin->phones GUI can BE an agent phone!). So after you have that set up properly ... you can test it with your cell phone as an agent and any web browser that can use the agent web interface.

Vicidial's settings are not standard. You may not see how it affects the system, but until you can initiate a vicidial agent session, it may be worth your while to try it and see if it helps. On the other hand, if the call initiates and succeeds all the way into a "You are the only person ...", you are "there" as it were. However, if you've placed the FQDN directly in the dial string, you may not be using the SIP context settings. Using the Global Variable with a "named context" which acquires the "host domain/ip" from within the sip context (recently acquired by name) ensures that your sip settings are indeed in use and therefore easily changed.

Remember that your knowledge of Asterisk MAY get in the way of a stock installation of a system designed to be installed by those who have NO knowledge of asterisk at all. This is why I suggest using the standard methods until AFTER your system is "UP". Then when you make changes, you'll know immediately if your change is why it isn't working (cuz it worked a second ago).

Most importantly: You MUST log on with a phone in Admin->Phones and hear "You are the only person in this conference". If this does not happen, you are NOT logged in properly as an agent, and the system will NOT work according to spec.

Happy Hunting!

[spinto]

I am able to login with an external phone and do hear "you are the only one on the conference", so I know the nexvortex.com account is working (otherwise, it would have never called me when I logged in). The problem is after I do a manual dial, the hangup button remains grayed out. It's as if VICIdial doesn't recognize that I'm in a call.

[williamconley]

ok, you have heard "you are the only person ..." and are logged in as an agent.

how do you "make a manual call" at this point? (precisely)

[spinto]

I logon to a campaign that's setup to manual dial. Once I logon and hear "You are the only person in this conference", I click the "MANUAL DIAL" link at the bottom of the screen. The Manual Dial screen appears and I enter phone number in the "Dial Override" text box (8xxxxxxxxxx). Here's the CLI of everything that's happening:


Logon to VICIdial
[Apr 7 21:20:30] == Parsing '/etc/asterisk/manager.conf': [Apr 7 21:20:30] Found
[Apr 7 21:20:30] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 7 21:20:30] -- Executing [817863389999@default:1] AGI("Local/817863389999@default-44de,2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 7 21:20:30] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 7 21:20:30] -- Executing [817863389999@default:2] Dial("Local/817863389999@default-44de,2", "SIP/17863389999@nexvortex.com||tTo") in new stack
[Apr 7 21:20:30] -- Called 17863389999@nexvortex.com
[Apr 7 21:20:33] -- SIP/nexvortex.com-00000009 is making progress passing it to Local/817863389999@default-44de,2
[Apr 7 21:20:36] -- SIP/nexvortex.com-00000009 is making progress passing it to Local/817863389999@default-44de,2
[Apr 7 21:20:40] -- SIP/nexvortex.com-00000009 answered Local/817863389999@default-44de,2
[Apr 7 21:20:40] > Channel Local/817863389999@default-44de,1 was answered.
[Apr 7 21:20:40] -- Executing [8600051@default:1] MeetMe("Local/817863389999@default-44de,1", "8600051|F") in new stack
[Apr 7 21:20:40] == Parsing '/etc/asterisk/meetme.conf': [Apr 7 21:20:40] Found
[Apr 7 21:20:40] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 7 21:20:40] Found
[Apr 7 21:20:40] -- Created MeetMe conference 1023 for conference '8600051'
[Apr 7 21:20:40] -- <Local/817863389999@default-44de,1> Playing 'conf-onlyperson' (language 'en')
[Apr 7 21:20:40] WARNING[3349]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Apr 7 21:20:40] -- Executing [h@default:1] DeadAGI("Local/817863389999@default-44de,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----0") in new stack
[Apr 7 21:20:40] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --10-----0 completed, returning 0
[Apr 7 21:20:40] == Spawn extension (default, 817863389999, 2) exited non-zero on 'Local/817863389999@default-44de,2'
[Apr 7 21:20:42] == Manager 'sendcron' logged off from 127.0.0.1

Make the call
[Apr 7 21:26:23] == Parsing '/etc/asterisk/manager.conf': [Apr 7 21:26:23] Found
[Apr 7 21:26:23] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 7 21:26:23] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-b1a7,2", "8600051|F") in new stack
[Apr 7 21:26:23] > Channel Local/8600051@default-b1a7,1 was answered.
[Apr 7 21:26:23] -- Executing [813054039999@default:1] AGI("Local/8600051@default-b1a7,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 7 21:26:23] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 7 21:26:23] -- Executing [813054039999@default:2] Dial("Local/8600051@default-b1a7,1", "SIP/13054039999@nexvortex.com||tTo") in new stack
[Apr 7 21:26:23] -- Called 13054039999@nexvortex.com
[Apr 7 21:26:24] -- SIP/nexvortex.com-0000000a answered Local/8600051@default-b1a7,1
[Apr 7 21:26:25] == Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

OK. And on that call you did not have a hangup customer button?

At the bottom of the agent screen, if you expand the conference link to show calls in the conference, do both calls show?

[spinto]

Here is what I see:

# REMOTE CHANNEL HANGUP VOLUME
1 Local/8600051@default-6ca3,2 HANGUP
2 SIP/nexvortex.com-0000000b HANGUP

When I hangup all is see is #2

[williamconley]

I need clarity.

I asked about having a hang up customer BUTTON.

you answered "when I hang up".

Please try again: On that call, did you have a hang up customer BUTTON (big red one, up top, in the agent screen area), when the two calls were both live?

Was the call successful? (Sound in both directions?)

[spinto]

I hungup the phone I manually dialed. The hangup customer button is disabled throughout the entire time I was logged on. Also, there is no "Live Call" at the top of the screen even though the called phone answers and a conversation takes place.

[williamconley]

Progress.

Now: When you expand below and see

Code: Select all

# REMOTE CHANNEL HANGUP VOLUME
1 Local/8600051@default-6ca3,2 HANGUP
2 SIP/nexvortex.com-0000000b HANGUP

Does that hangup button next to each channel work? (ie: if you press either hangup button, does it hang up that channel next to it successfully?)

[spinto]

It hangs up with no problems and I get the pop-up saying the hangup command sent ....

[williamconley]

Then there is a mere challenge of how Vicidial is interpreting and recording the agent and/or client phone landing in the meetme room.

now would be a good time to either change to "standard" with the globals variable (et al) or reinstall the system (clean) since you did not install it (and you have no idea what weirdness may have occurred prior to now) or begin digging in to find out what would cause Vicidial to not realize the calls are in the room.

Primarily, I'd be looking at why the agent phone is in the room as "local", which it really should not be.

Also: Have you tested this without the override? How about Autodial mode?

[spinto]

I just tried it by adding a lead, then manually dialing the lead (not manual override) and it worked. I also tried it predictively and it worked. This should help me figure out why the manual dial override isn't working.

[williamconley]

Or as they say on HeeHaw ... don't do it that way! LOL

You don't need "override" to use "Manual Dial" and manually enter a lead.

[Alex]

Hello William,
I see that the 'Test Calling Disabled' in Campaign (At the bottom of the page). How do I enable it please?
Also I'm unable to do 'Dial Override' in Manual Dial section for test call without generating the lead, but no action at all. It directly goes back to customer info screen with pause state. By entering the number in Phone Number field in manual dial screen I'm able to make call.

Please advise.

Thanks in advance.

[williamconley]

Hello William,
I see that the 'Test Calling Disabled' in Campaign (At the bottom of the page). How do I enable it please?
Also I'm unable to do 'Dial Override' in Manual Dial section for test call without generating the lead, but no action at all. It directly goes back to customer info screen with pause state. By entering the number in Phone Number field in manual dial screen I'm able to make call.

Please advise.

Thanks in advance.

You seriously need to do two things:

1) Read the Vicidial Manager's Manual (in your case I would recommend the paid version, and NO I don't have anything to do with the company selling it ... I'm just saying that you sound like someone who desparately needs it to avoid paying others for what you could get in the manual, at least the free version). go cover to cover. Seriously.

2) Explain why you are attempting each of these interesting tasks. Why you think you need to do them. Also explain your settings relevant to the task at hand, what you think should happen, and what really happens. If there's any Asterisk CLI output, include it (not 3000 lines of unrelated code, but definitely everything involving the phone call in question from inception up to your problem). And put each task in its own post with a suitable subject line instead of diving in on someone else's "almost similar but not really related" existing post. That way your subject line may appeal to some other future technician asking the same question, and help them find the same answer while keeping the discussion on each topic segregated which will reduce confusion.