FAQ

What is the minimum recharge amount?

$50

What payment types are accepted?

What payment types are accepted?
We accept Bitcoins, Payoneer, Payza, PayPal, Master Card, Visa Card, American Express, MoneyGram and Wire Transfer. All the data (password, card number etc.) used while doing payments via Master Card, Visa Card and American Express are fed into the PayPal system and checked out by PayPal.

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

What is the minimum recharge amount?

$50

Sample of eyeBeam configuration

SIP Accounts

Enable this SIP account – check this box
Display Name – YOUR_USERNAME
User name – YOUR_USERNAME
Password – YOUR_PASSWORD
Authorization user name – YOUR_USERNAME
Domain – Host IP
Register with domain – check this box

All other settings leave as default.

How do I troubleshoot a 401/403 Authentication Error?

Authentication Errors like the 401 or 403 mean that the server is rejecting your connection, usually for Registration, but sometimes for calling.
Typically these errors are caused by incorrect Account settings, usually username and password. Check these again with your VoIP provider and be sure the account is correctly configured.
Sometimes a server controls where a user may connect from – i.e. the local network, but not the Internet. If you can connect in some places but get the Authentication Error in another, this may be the case. Contact your Server Administrator about this.
Sometimes these messages are accompanied by additional text explaining the error – if so, ask your VoIP provider or Server Administrator about these errors.

What does error 407 – Proxy Authentication Required mean?

Error ‘407 Proxy authentication required’ indicates that the client must now authenticate itself with the proxy. this error is normal in networks where authorization is enabled (most networks) and can occur once per SIP request and is generated by the network. If you are continuously getting this error, either you have entered incorrect login information (username/password), or your account has not been provisioned or set up properly.
If you are using one of our retail products, verify that your account credentials have been properly entered. If the problem persists, we recommend that you follow up with your VoIP service provider.

What does error 404 – Not Found mean?

Error ‘404 Not Found’ indicates that the server had definite information that the user does not exist in the specified domain. We recommend that you ensure you have entered a valid username or telephone number and try again.

How do I troubleshoot a 500/502/503 error when registering or calling?

Error ‘503 Service unavailable’ indicates that the server or gateway is unable to process the request due to an overload or maintenance problem.
The error may also say “Internal Server Error.”
Make sure your transport and encryption settings are correct.
We recommend that verify your account settings and if the problem persists to contact your VoIP service provider.

403 Forbidden

It might be the codec issue as well if its intermittently giving you the 403 Forbidden. If it’s not recognizing the proper codec for VoIP, it might give you that error. I tried unchecking all other codecs except G729 or G711u and it fixed the issue for me.
Eyebeam Re-registration Fails Occasionally
Some of you might be using eyebeam and noticed that when it reregisters sometimes it fails. Now I have sent some traces to eyebeam about this problem and the response was as follows:
In reg1 the proxy sets a timer of 60 seconds for the initial (successful) registration (packets 45-46). 55 seconds later eyeBeam tries to re-register, but the proxy returns a 403 Forbidden (packet 1511). My guess is the proxy isn’t willing to accept re-registration until the full 60 seconds expire.
I changed my eyebeam setting to 65 seconds for reregistration and it seems much better. Note that this was an intermittent issue, eyebeam does not seem to attempt this all the time but this setting seems to make it far more stable.
For voipfone, your reregister setting should be 60 seconds at all times, the re-register of 3600 will cause problems such as this. Please change it to 60 seconds and see what happens. If you still have problems let us know.

Service Unavailable

That error usually occurs because of a network issue, such as a router/firewall device that may block packets after a period of time of no activity.
In eyeBeam under Account settings|Advanced tab, try reducing the re-register setting, and check “Send SIP keep-alive”.That will help with re-establishing the connection to the server.

Registration Error, 408 – Request TimeOut

If you are getting a 408 error from x-lite this means you are not receiving any response from the sip registration server that you are attempting to connect to with x-lite. There are a number of reasons for this but they could be caused by:
a) Your router b) Your firewall c) Anti-spyware
Try disabling any firewall software you have running on your desktop. You can also try connected directly to your cable/DSL modem and bypass your router to determine if the router is causing this problem. You could also try disabling the QOS setting which is under Options -> Advanced Menu -> Quality of Service.
This error may be caused by incorrect server information (domain, server address) entered in your SIP account. Double-check this information.
It may also be caused by a bad or non-existent network connection. Check your device to be sure it’s properly connected to WiFi or your Mobile Network.
If this is your first time connecting and the above haven’t helped, try another network connection to see if your first connection might be at fault.

How to Increase & Decrease Call Flow (all the options)

How to Upload Leads

How to create campaign

At a time how many campaign we can run?

what is the remote agents – number of lines function?

What is remote agents & what purpose we use it?

Sorry, your phone login and password are not active in this

Check the following settings. Navigate to Telephony > Users > Choose the user > check if your phone login password matches with the Phone extension login password. Phone extension login password can be found by navigating to Admin Settings > Phones

Agent XXX is currently in use or improper logout. Contact Administrator. Follow the steps below:

1. Go to Telephony.
2. Choose and click the USERS.
3. Select the Agent that having a problem.
4. Click the GREEN button (Info user AgentXXX).
5. Click FORCE LOGOUT.

I get “Sorry, there are no available sessions”. What do I need to do?

Make sure in the admin section under admin->conferences->Vicidial Conferences your server IP is defined/displayed. If you have re-configured your network settings, make sure the new IP address is applied to your Vicidial configurations. Run the following command:
update_server_ip

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

I am getting one-way or no audio on my calls. Why is that?

These problems are normally related to firewall/NAT issues. If your GOautodial/VICIdial server is behind a firewall, edit sip.conf:
nano /etc/asterisk/sip.conf
Replace:
;externip = 192.168.1.1
to this:
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
asterisk -rx “reload”

How do I update my system?

Just run the following command:
yum update -y This will download and install the latest system updates. Updating the system regularly is recommended. Bug fixes and security patches are applied.

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

What about hardware? How do I know if a particular NIC or motherboard is compatible?

GOautodial is built on CentOS which is itself based on Red Hat Enterprise Linux. The current version of GOautodial uses CentOS 5 as it’s base. Red Hat has a hardware compatibility list (HCL) for versions 3, 4 and 5 here: https://hardware.redhat.com/

What T1/E1/Analog telephony cards do you recommend?

GOautodial is tested with Sangoma and Digium. It has out of the box support for the two. The important thing to remember is that as long as it works with Asterisk, it will work with GOautodial.

Does GOautodial work with trunks other than SIP?

Yes. GOautodial also works with IAX, Analog and E1/T1 lines. It utilizes trunks being used by Asterisk. H.323 should also work but we haven’t fully tested it and its also not installed and configured by default.

What phones will work with GOautodial/VICIdial?

Most any SIP compatible phone from companies like Aastra, Polycom, Linksys, SNOM, Cisco, and others will work, you want to make sure it is fully SIP compliant. You can also use a regular analogue phone if you have a card with an FXS port on it or you can use an ATA (analogue telephone adapter) to bridge between SIP and the analogue phone. As long as it works with Asterisk, it will work with GOautodial.

Is GOautodial free?

Yes.

What is the minimum recharge amount?

$50

How to Increase & Decrease Call Flow (all the options)

How to Check Reports

How to Check Live Call?

How to Upload Recordings?

How to Upload Leads

How to create campaign

At a time how many campaign we can run?

what is the remote agents – number of lines function?

What is remote agents & what purpose we use it?

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

I am getting one-way or no audio on my calls. Why is that?

These problems are normally related to firewall/NAT issues. If your GOautodial/VICIdial server is behind a firewall, edit sip.conf:
nano /etc/asterisk/sip.conf
Replace:
;externip = 192.168.1.1
to this:
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
asterisk -rx “reload”

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

What phones will work with GOautodial/VICIdial?

Most any SIP compatible phone from companies like Aastra, Polycom, Linksys, SNOM, Cisco, and others will work, you want to make sure it is fully SIP compliant. You can also use a regular analogue phone if you have a card with an FXS port on it or you can use an ATA (analogue telephone adapter) to bridge between SIP and the analogue phone. As long as it works with Asterisk, it will work with GOautodial.

Is VICIdial free?

Yes.

Is GOautodial related to the Vicidial group?

No! GOautodial is in no way related to the Vicidial group.

Is GOautodial the same as VicidialNOW?

Yes. We renamed the project to “GOautodial” since the word “Vicidial” is a registered trademark. The name change was necessary since GOautodial evolved from being more than just a Vicidial distribution. It’s now a complete open-source dialer system.

How to Increase & Decrease Call Flow (all the options)

How to Check Reports

How to Check Live Call?

How to Upload Recordings?

How to Upload Leads

How to create campaign

At a time how many campaign we can run?

what is the remote agents – number of lines function?

What is remote agents & what purpose we use it?

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

How to Increase & Decrease Call Flow (all the options)

How to Check Reports

How to Check Live Call?

How to Upload Recordings?

How to Upload Leads

How to create campaign

At a time how many campaign we can run?

what is the remote agents – number of lines function?

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

How to Increase & Decrease Call Flow (all the options)

How to Check Reports

How to Check Live Call?

How to Upload Recordings?

How to Upload Leads

How to create campaign

At a time how many campaign we can run?

What is remote agents & what purpose we use it?

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

I am getting one-way or no audio on my calls. Why is that?

These problems are normally related to firewall/NAT issues. If your GOautodial/VICIdial server is behind a firewall, edit sip.conf:
nano /etc/asterisk/sip.conf
Replace:
;externip = 192.168.1.1
to this:
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
asterisk -rx “reload”

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

Why am I getting “choppy” calls? Why are most of my calls of poor quality? Are you inside a tunnel?

There are a lot of factors affecting the quality of calls. They are mainly:
Asterisk codec being used by the server
Agent workstation
Bandwidth consumption
Overloaded workstation
Softphone (try to use other softphones like zoiper, xlite and eyebeam)
Poor quality headset (USB headsets are highly recommended)
If you have limited bandwidth, the codec used by your GOautodial/VICIdial server (to your SIP gateway) should either be GSM or G729. These are bandwidth efficient codecs.
You might also need to check the agent’s workstations to see if they’re not overloaded. Meaning they’re just running the necessary applications for dialling (Firefox, softphone, notepads or etc). Softphones eat CPU resources. If the workstation is overloaded then call quality can suffer.
Lastly, check your bandwidth consumption. You might be eating all your bandwidth. Make sure your internet connectivity is just being used for dialling purposes. Browsing social media sites like Facebook, Google+, Youtube and others will eat up most of your bandwidth.

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

I am getting one-way or no audio on my calls. Why is that?

These problems are normally related to firewall/NAT issues. If your GOautodial/VICIdial server is behind a firewall, edit sip.conf:
nano /etc/asterisk/sip.conf
Replace:
;externip = 192.168.1.1
to this:
externip = 192.168.1.1
Where 192.168.1.1 is your public IP address. Reload Asterisk after the changes.
asterisk -rx “reload”

Help! All my RAM is being eaten up! What do I do?

Probably nothing. The short answer is that Linux (and other Unix like systems) use RAM differently than you may be used to in other operating systems. The long answer is outside the scope of this FAQ, but the following links have more information: http://www.linuxhowtos.org/System/Linux%20Memory%20Management.htm

Sample of eyeBeam configuration

SIP Accounts

Enable this SIP account – check this box
Display Name – YOUR_USERNAME
User name – YOUR_USERNAME
Password – YOUR_PASSWORD
Authorization user name – YOUR_USERNAME
Domain – Host IP
Register with domain – check this box

All other settings leave as default.

What is the validity of the VoIP account?

It’s for life time.

What is the support for recharge and troubleshooting?

You can recharge anytime by contacting on Skype: voip.dialer4 or our LIVE CHAT option in the website.

Can I register online for the voip account?

Yes, you can! Click on the link Sign up and fill-up the form to get the account instantly.

What is the minimum recharge amount?

$50

Can I test your route or can I get a demo account?

Yes, you can test our route/minutes with a free credit of $1. Kindly either sign up or use the demo account details to configure with your softphone for the test. Links are available on the home page.

Can I check call details online?

Yes, you can access your Call Details report online.