Looking for support – GOAUTODIAL set-up

[SISTA India]

Hi All,
I am very new to call center mechanism and trying to understand and learn in and out of it. I am setting up a call center in Bangalore, India with 15 agents.
I have installed the goautodial few weeks back and configure the entire component as per “start-over guide” as well as “forum post”.
But still I have some queries, like;
- [b]When I am logging – in as agent [/b]I am able to hear “You are currently the only person in this conference” but after that I am not able to hear any Voice like “dial tone or something like that”
- I am not able to make any manual call through my soft phone (X-lite -3)

Please help me with your expert advice, so I can come out with this problem and able to enjoy this superb dialer.
Thanking you with anticipation
Regards,
Mitesh

[covarrubiasgg]

Hi Mitesh welcome to the forum,

Once you are logged into the system you will not hear any dial tone or something until you start manually dialing or if it is predictive dialing until a call is sent to the agent.

When you said you cant make manual calls from your softphone, you mean call trought a VoIP service or calls extension to extension?

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

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.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

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)

[SISTA India]

Dear covarrubiasgg,

Thanks for your quick reply.

I tried to make call with goautodial’s manual dialing settings also, but still I am not able to hear any sound.

When I configure my X-lite phone directly with “username, password and domain” provided by my Voip service provider I am able to call manually, but when I configure the same with “Goautodial” credentials I am not able to make call manually. It says “call failed: Not found”

Please suggest

Regards,

Mitesh

[covarrubiasgg]

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

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.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

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)

Did you register your server with you provider first, there are some steps you should complete before start placing calls, try to get the vicidal Managers Manual from the vicidial store:

http://www.vicidial.org/store.php

There you will find a very well explined tutorial on how to setup a new carrier, and how to place calls using your vicidial server.

[williamconley]

I tried to make call with goautodial’s manual dialing settings also, but still I am not able to hear any sound.

When I configure my X-lite phone directly with “username, password and domain” provided by my Voip service provider I am able to call manually, but when I configure the same with “Goautodial” credentials I am not able to make call manually. It says “call failed: Not found”

Please AT THE VERY LEAST post your installation method and Vicidial Version with Build.

Your installation method would be the .iso you downloaded (the name of the file you downloaded works nicely) and the version of it should also be included (don't leave that part out to shorten your post).

Your Vicidial Version with Build is available at the bottom left corner of almost all Administrator screens.

Ordinarily, without this information there is not a lot of help we can offer. In addition to that, it is a requirement to use this cool free support forum. So it is handy for a lot of reasons.
___

When you build a Vicidial system, your system must have a carrier installed to be able to make calls. Any phones you "register" to your vicidial system will then be dialing to those carriers through vicidial. But if vicidial doesn't have a carrier set up ... obviously that won't work.

So ... since you've read the manual and worked through the tutorial on how to set up your vicidial system, can you tell me where in the manual you stopped or skipped a section

That's generally the problem, go back and do all of it, and your system will work unless you have a "problem", in which case you come here ... tell us where your problem occurred That's what we're here for. You give us the page number/exercise and the "problem" you encountered and we help you through it.

[SISTA India]

Hi,
I downloaded the ISO from goautodial website and the components are
- CentOS 5.5
- VICIDIAL 2.2.1
- Asterisk 1.4.27.1-vici
- vTigerCRM CRM 5.10

I followed the instructions given in “vicidialnow-getting-started-guide – wiki”

Following is my CARRIER set-up

Registration String -
register => 18826633333:194217510#@85.232.50.152

Account Entry:

[VIVA Com]
disallow=all
allow=ulaw
allow=alaw
type=friend
secret=194217510#
username=18826633333
host=85.232.50.152
context=default
qualify=yes
insecure=very
nat=yes
formdomain=85.232.50.152

Globals String: TRUNK = SIP/VIVA Com

Dialplan Entry

exten=>_91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXXX,3,Hangup

exten=>_91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXXXX,3,Hangup

exten=>_91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXX,3,Hangup

and I configure my X-lite phone with following credential

Display Name: cc101
User Name: cc101
Password: test
Domain: 192.168.1.2

Same time please let me know -
How to assign extension no. to agents system and how many g729 do I need for 15 agents set up.(I am going for free g729)

Regards,
Mitesh

[covarrubiasgg]

are you registred with you carrier?

Code: Select all

sip show registry

Can you post your CLI with your dial process so we can check what are you sending to your carrier and what response you are getting from them?

[williamconley]

exten=>_91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXXX,3,Hangup

exten=>_91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXXXX,3,Hangup

exten=>_91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA Com,,To)
exten=>_91XXXXXXXXX,3,Hangup

1) you did not actually tell us where you left off or had your first problem in the manual. IF you used only the goautodial manual, and you did not find a tutorial to set up for your first outbound call, try the Vicidial manual and do the tutorial in that (from eflo.net, free version available if you're on a tight budget).

2) you set up your global variable, then did not use it! LOL (not to mention the missing dot in VIVA.com, but that could merely be how you got it past the phpBB filter i suppose). The global variable belongs after the @ instead of VIVA.com. Since you skipped that, the system has no idea which protocol you are using (you must include SIP/ which is part of the global variable definition). This is one of the reasons to use the global variable, it simplifies entry.

3) The results should be much more visible if you post your CLI, but also tell us where you were in the manual(s) so the creator(s) of the manual(s) can modify anything that confused you. So all the noobies can learn and not get stuck here.

[SISTA India]

Hi,

I am very new to call center mechanism and trying to understand and learn in and out of it. I am setting up a call center in Bangalore, India with 15 agents.
I have installed the goautodial few weeks back and configure the entire component as per “start-over guide” as well as “forum post”.

I downloaded the ISO from goautodial website and the components are
- CentOS 5.5
- VICIDIAL 2.2.1
- Asterisk 1.4.27.1-vici
- vTigerCRM CRM 5.10

I followed the instructions given in “vicidialnow-getting-started-guide – wiki”

Following is my CARRIER set-up

Registration String -
register => 18826633333:194217510#@85.232.50.152

Account Entry:

[VIVA]
disallow=all
allow=ulaw
allow=alaw
type=friend
secret=194217510#
username=18826633333
host=85.232.50.152
context=default
qualify=yes
insecure=very
nat=yes
formdomain=85.232.50.152

Globals String: TRUNK = SIP/VIVA

Dialplan Entry

exten=>_91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA,,To)
exten=>_91XXXXXXXXXX,3,Hangup

and I configure my X-lite phone with following credential
Display Name: cc101
User Name: cc101
Password: test
Domain: 192.168.1.2

After completing everything according to guide I uploaded some 20K calling data (Canada), but when I started dialing the status always shows “BUSY” around 17K data status is coming same as “BUSY”, and its same for predictive as well as manual both.

But when I configured X-lite with my VOIP provider credential (without goautodial) I am able to connect with almost all the calls.

I tried all possible ways to fix the dialer but I am not able to do so. It’s the third time I am going to re-install the dialer.

Please help me with your expert advice, so I can come out with this problem and able to enjoy this superb dialer.
Thanking you with anticipation.

[williamconley]

1) does your sip trunk register to viva?

Code: Select all

sip show registry

2) you created your globals string, but did not use it.

Globals String: TRUNK = SIP/VIVA

this defines "TRUNK" as a global variable to be used in dial plans

exten=>_91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@VIVA,,To)

here is the dial plan, but you used VIVA instead of TRUNK. So the system should just go to VIVA, right? But where is VIVA? Is it SIP or IAX2? Since you did not use TRUNK, which is "SIP/VIVA" and only used "VIVA", the SIP/ was left off entirely, so no protocol is specified and the system cannot dial. Change to:

exten=>_91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@TRUNK,,To)

but be aware that TRUNK may be defined elsewhere (I usually use DIAL9TRUNK for both the globals string and the exten entry, which is definitely NOT defined anywhere else. And of course, I would then have DIAL8TRUNK etc available for definition for other carriers.
3) If you were to show us your asterisk CLI from a call attempt, it would be helpful resolving your problem. The concept is this: if you call your mechanic and tell him your car doesn't work, and then list EVERY part in the car, he will still have no idea what is wrong with the car (even if a part is wrong, that's an awful long way to go instead of just letting him listen to the engine or touch the car ...). I can't drive over there and look at your server, but you can let me watch the telephony engine ... it's the asterisk CLI. This gives other users the ability to see the inner workings and render an opinion of what is wrong in this next phase.

[SISTA India]

Dear William,
Thanks a lot for your support

As I mentioned I am very new with this mechanism and trying to understand and learn in and out of it. As you ask to show asterisk CLI from a call attempt, please let me know how to access asterisk CLI from server so I can share it with you.

Same time please help me in understanding that, which dialer will be more beneficial in long run (goautodial or vicibox)

[williamconley]

Code: Select all

asterisk -R

then

Code: Select all

core set verbose 20

Both GoAutoDial and Vicibox install VICIdial (which is the dialer software) and asterisk (the telephone engine) and all necessary support software (web server, database server, etc).

Operating System: GoAutoDial installs CentOS (A Redhat related OS). Vicibox Redux installs OpenSuSE.

Vicidial Version: GoAutoDial installs 2.2. Vicibox can install 2.2 or 2.4 (chosen during installation).

Pros of GoAutoDial: Pretty Agent Screen. Pre-installed Trunk (purchasing minutes through Gardo).

Pros of Vicibox: Supported by/Created by The Vicidial Group (The makers of VICIdial!).

Cons of GoAutoDial: CentOS (and Redhat in general) is not supported by The Vicidial Group. I only recommend this installation to clients who MUST have CentOS or who really like the Pretty Screen for the agents.

Cons of Vicibox: No pretty screen. No pre-installed trunk.

Universal:

Either version can be upgraded to the latest version of Vicidial (but you lose the pretty agent screen if you were using it because it is not compatible with the latest version of Vicidial). But neither installation has a "limited lifespan".

Either version CAN use the pretty screen (but on Vicibox you must be sure to install 2.2 and then you must manually copy the files for the "agc2" folder to use this screen). I don't actually know anyone who has done that mind you, but it is a GNU screen just like the rest of the system.

Both have all the scripts for backup and other functions.

Databases backed up on one can actually be used on the other (as the database is the same on either system).

Really they are the same, with those minor differences (depending on how much importance you place on support by The Vicidial Group, and the other differences).

[SISTA India]

Hi,
My sip trunk is register and even my extension is also showing register.
I tried all possible way to rectify the problem, even I reinstall and reconfigure it 3 times from Friday, but I am facing the same problem.
Every time I am trying to make a call I am getting the same massage
“Call Rejected :BUSY”
Cause: 21 – Call Rejected

Please help

[williamconley]

i asked for cli output ... you asked for the method to obtain cli output. then you didn't post any cli output. can't help without the cli output.