Quick softphone question?

[aemae]

If when I log in to the agent log in, my softphone rings and I get the only in conference message, but it still gives the "no one is in this session 8560001 or whatever it is, could I assume that my CARRIER PHONE and SOFTPHONE are set up correctly and the problem lies elsewhere?

Thanks Eric

[williamconley]

Unless you can give the exact error message and (of course) tell us where you got the message in question, it's not likely we'll be able to resolve the issue.

Excellent job posting your specs, by the way. But it almost seems like you are "remembering" an error that "may" have happened or perhaps relating something someone else told you ...

First step is always to reproduce the error and get the actual problem locked down.

[aemae]

The exact error message used to be no one is in your session, but now when I click dial next customer, it waits for ring then delivers this error:

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Channel unavailable Cause 16 Normal Call Clearing   

What does that mean? How do I fix it? I know you may want the CLI i think it is called or Asterisk error sheet, however I do not know how to get those... I am only able to log in to a workstation via team viewer and then log in to vici by using the server address. But I am getting close!! I receive the call upon agent log in, which I think means carrier settings are alright right?

[williamconley]

How does it "deliver" this error? I presume you mean this is what is shown on the agent screen in a javascript popup?

If so, and your phone DID ring, then it is likely necessary to use PuTTY (or another ssh terminal application, putty is available from chiark as a simple .exe downloadable app) to access the asterisk CLI and determine why the call terminated.

After you log on to the server, the asterisk cli is accessed via:

asterisk -R

Perform a single test call and read the result. It's a little "matrix-like", but it does have english in there.

It's also quite likely that you just have not configured your carrier properly. Post your Carrier settings (xxx/yyy/zzz for user/pass/ip to avoid posting anything personal). There are hundreds of posts showing various proper carrier setups, of course, such as http://www.eflo.net/VICIDIALforum/viewt ... 510#p63510

[aemae]

http://www.vicidial.org/VICIDIALforum/v ... =4&t=26901

My carrier settings are shown in I think the 5th post from the bottom. I basically used what was provided to me, from my CARRIER website which is

http://www.switch2voip.us/index.php/sup ... n-vicidial

[williamconley]

which brings us back to asterisk CLI output. while there are a lot of mechanics who can diagnose any problem with your car without ever opening the hood ... they are generally costly and disreputable. need that look under the hood.

[aemae]

How do I go about getting the CLI output? Do i install something on the server? The server just has a black screen so I would not know how to go about installing anything on it, but I realize now that the CLI output is the descriptive error message we need to diagnose where the issue is. So thank you very much. I feel it getting closer!! Also just fyi I used to work for a company, they are one of the six aggregators of data in the us, so if you or anyone you know uses data lists for ur dialers I can get ridiculous discounts and even a white label platform for your site! They are not resellers only bigger company is infousa.

[williamconley]

Linux CLI is available either through the console directly (by sitting at the machine) or by using a terminal application. My favorite is PuTTY (by chiark ... google those two words and you'll get it). Just download the .exe file from the chiark site, no installation necessary. Just run it.

Then you'll be at the linux CLI without having to visit the machine itself. Better yet: you can now copy/paste to and from the machine with your Windows text (so you can copy a command entered here into PuTTY without having to retype it ...).

Once in the linux cli (console or PuTTY, either one):

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asterisk -R

This will show the command line output from asterisk where you can watch calls progress.

Happy Hunting

[aemae]

Cool! thank you very much. I ran into a problem, when I type the command on the console it asks me for a password... I assume this is the server password which to my knowledge i never changed, and would still be "vicidial". Which password is this referring to? and where would i find it if i forgot it?
thanks

Also when I try to connect remotely I am running into the same problems listed in another thread. I understand I must open some ports I believe either on the office side or both the office and home. Which ports do I have to open and on which side, and also pointed to where? I am sorry for the ignorance but ports are something I haven't messed with really. I only want one computer to be able to connect, and preferably I'd like it to be from anywhere, is there a way to allow only a certain computer in no matter the IP or location?

[williamconley]

port 22 is the standard ssh port. be careful opening it as there are hackers watching. only open it for your IP or close it as soon as you are done. also iptables is a security application installed in Vicibox which can be used to limit access (yast firewall ... then look for "custom" in the left pane an you can add individual IP address for external access). but you must first remove all other access. it's not hard, and there have been several posts for iptables whitelisting as well as our dynamic good guys app.

your root password is required to log in. asterisk -R requires no password, but you must already be logged in to use it. there is no known method to acquire the root password if you lost it, but it can be reset (there are posts available for this as well). the reset of a root password in OpenSuSE is covered on many other forums, however, and it requires physical access to the machine in question.

[aemae]

Code: Select all

login as: root
Using keyboard-interactive authentication.
Password:
Last login: Wed Feb 13 14:47:04 2013
Thank you for installing ViciBox Server v.4.0!
This software is available for free download at
http://www.vicibox.com. If you paid for this
software you have been ripped off. Please report
any fraud or abuses of this software to
abuse@vicidial.com. Please report any bugs on
the forum at http://www.vicidial.org

To configure the LAN settings type:
yast lan

To change the server IP in the database type:
/usr/share/astguiclient/ADMIN_update_server_ip.pl

Official paid-for ViciDial support is available at
http://www.vicidial.com

Free community-based ViciDial Support is available
at http://www.vicidial.org/VICIDIALforum

- ViciBox Redux v.4.0.2 released
linux-pjhtSolar:~ # asterisk -r
Asterisk 1.4.44-vici, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.44-vici currently running on linux-pjhtSolar (pid = 2270)
Verbosity is at least 21
[Feb 13 15:12:01]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:01] Found
[Feb 13 15:12:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:01]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:01] Found
[Feb 13 15:12:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 13 15:12:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 13 15:12:03]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:03] Found
[Feb 13 15:12:03]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:06]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:06] Found
[Feb 13 15:12:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 13 15:12:09]        > Channel SIP/123-00000000 was answered.
[Feb 13 15:12:09]     -- Executing [8600051@default:1] MeetMe("SIP/123-00000000", "8600051|F") in new stack
[Feb 13 15:12:09]   == Parsing '/etc/asterisk/meetme.conf': [Feb 13 15:12:09] Found
[Feb 13 15:12:09]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Feb 13 15:12:09] Found
[Feb 13 15:12:09]     -- Created MeetMe conference 1023 for conference '8600051'
[Feb 13 15:12:09]     -- <SIP/123-00000000> Playing 'conf-onlyperson' (language 'en')
[Feb 13 15:12:11]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:11] Found
[Feb 13 15:12:11]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:11]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-d0ee,2", "8600051|F") in new stack
[Feb 13 15:12:11]        > Channel Local/8600051@default-d0ee,1 was answered.
[Feb 13 15:12:11]     -- Executing [918057780747@default:1] AGI("Local/8600051@default-d0ee,1", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 13 15:12:11]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 13 15:12:11]     -- Executing [918057780747@default:2] Dial("Local/8600051@default-d0ee,1", "SIP/18057780747@Switch2Voip||o") in new stack
[Feb 13 15:12:11] WARNING[2750]: chan_sip.c:3346 sip_call: No audio format found to offer. Cancelling call to 18057780747
[Feb 13 15:12:11]     -- Couldn't call 18057780747@Switch2Voip
[Feb 13 15:12:11]   == Everyone is busy/congested at this time (0:0/0/0)
[Feb 13 15:12:11]     -- Executing [918057780747@default:3] Hangup("Local/8600051@default-d0ee,1", "") in new stack
[Feb 13 15:12:11]   == Spawn extension (default, 918057780747, 3) exited non-zero on 'Local/8600051@default-d0ee,1'
[Feb 13 15:12:11]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d0ee,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Feb 13 15:12:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 13 15:12:11]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL---------- completed, returning 0
[Feb 13 15:12:11]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d0ee,2'
[Feb 13 15:12:11]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d0ee,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Feb 13 15:12:11]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Feb 13 15:12:13]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 13 15:12:25]   == Parsing '/etc/asterisk/manager.conf': [Feb 13 15:12:25] Found
[Feb 13 15:12:25]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 13 15:12:27]   == Manager 'sendcron' logged off from 127.0.0.1
linux-pjhtSolar*CLI>


This is my CLI from the point of me logging in as an agent, answering softphone, and attempting to dial a number by clicking "dial next number". On the agent screen is says waiting to dial, and counts to 10, then says Channel unavailable cause: 16

If you could identify my problem it would be great, thank you very very much!! loveyouwill#

[williamconley]

Well posted. A little squishy in the emoticon area, though. LOL

Code: Select all

No audio format found to offer.

I'm gonna say ... it has no codec to use. This would likely be in your Account Entry for the carrier. It usually expects something like "allow=ulaw".

[aemae]

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[Switch2Voip]
type=peer
username=XXXXXXXXX
host=sip.switch2voip.us
fromuser=XXXXXXXXX
canreinvite=no
reinvite=no
secret=XXXXX
disallow=all
allow=g729

It shows "allow=g729", and I copied it directly off the site that basically tells you what to put in for vicidial. The site can be found here:

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http://switch2voip.us/index.php/support/how-to-setup-your-trunk-in-vicidial

Should I try a different codec? Or maybe I dont have it configured properly to use the specified "g729" codec? Also what is the deal with 'manager sendcron' logging in an out? It pops up alot on the CLI.

[williamconley]

configuring for g729 is marvelous. IF (and only IF) you have installed g729 and it is working and the carrier supports it.

before you go to g729, however, it is best to get it all working with ulaw. remove =g729, add =ulaw. work out g729 ... later

[aemae]

Will, you are the man. The ulaw codec worked fine, and I am now making predictive calls and making disposition. I have more questions (yay).

My next task will be setting up inbound calling. How does vicidial determine which agent to connect the call too? I'd like to make it where the correct customer goes to the correct agent. Does this mean I would set up an inbound group for each agent?
Would this mean each agent would have their own caller id, and I would maybe have one universal inbound group with a recorded message and IVR to route correctly?

Also it seems there is no way to redial a customer if they disconnect. In telemarketing my signature trademark is calling back the hang ups and then converting to a sale. I would like my agents to be able to do the same. How is this achieved?

Third question is, how do I know how many channels I can use? Also is there a way to see how many channels are being used? I only have 2 agents on the dialer right now, basically testing. It seems they are waiting between 15-30 seconds between getting a call, or answering machine. Is this typical? Is there a way to speed it up?

Sorry for the SAT type short answer questions lol
Thanks Will!

[williamconley]

1) define "correct customer" and "correct agent". LOL (are you saying you have an agent named 'correct' and he should only accept phone calls from prospects if they share his name? unlikely ... but it's the only way that description works out as anything beyond "vague" ... )

2) Inbound is covered in the manager's manual. Which I'm sure you have completed from cover to cover ... right?

3) If manual dialing and view call log are both enabled for users, they can see their calls and call them back

4) Ask your provider how many channels you can use. Vicidial maxes out around 150-350 per server depending on configuration, usage and Power. This is controlled in Admin->Servers on a per server basis ... but beware setting it too high without discussion with your provider. They Will cut you off if you exceed your limit. There's always a limit. Many start with as low as 10 channels.

5) Dial Ratio 2.5 start. Increase or decrease carefully while watching the Real Time monitor. Be sure you watch for at least 10 minutes before deciding it was a good change. And turn on "Show Carrier Stats" in " Choose Report Display Options " so you can see if you are getting too much congestion (rejected calls).