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Is bandwidth usage calculated from outbound, inbound, or inbound+outbound traffic?

Your bandwidth usage is based on the highest of either your inbound traffic or your outbound traffic. For example, if your VPS uses 100 GB of incoming bandwidth and 200 GB of outgoing bandwidth, your utilization for billing purposes would be 200 GB.

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.

What does error 404 – Not Found mean?

Error ‘404 Not Found’ indicates that the server has 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.

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.

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 the 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.

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.

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 several 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 connecting directly to your cable/DSL modem and bypassing 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.

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.

Sample of eyeBeam SIP/VoIP user based account configuration

👉 Select –> SIP Accounts Settings

👉 Select –> Add

👉 Enter the value as mentioned below:

Display Name – YOUR_USERNAME
User name – YOUR_USERNAME
Password – YOUR_PASSWORD
Authorization user name – YOUR_USERNAME
Domain – Host IP

👉 Domain Proxy

👉 Check this box: Register with the domain and receive incoming calls

👉 Send outbound via:

👉 Select this: target domain

👉 Click Ok

You have successfully configured the SIP/VoIP account on eyebeam

Note: This is only for a user based SIP account and not for an IP based.

What does 6/6 & 1/1 pulse mean (e.g., USA $0.0085/min 6/6 pulse & UK $0.0095/min 1/1 pulse)?

In VoIP (Voice over IP) billing, a pulse refers to the billing increment — how the duration of a call is measured and charged. It affects how the rate per minute is applied to the actual call time.

E.g., USA $0.0085/min 6/6 pulse

The initial billing block is 6 seconds.

Every subsequent block is also 6 seconds.

1–6 seconds, they’re charged for 6 seconds.

7–12 seconds, they’re charged for 12 seconds.

And so on.

This means you’re always billed in multiples of 6 seconds, regardless of how much of that block you use.


E.g., UK $0.0095/min 1/1 pulse?

Calls are billed per second, starting from the first second.

No rounding or minimum blocks—just pure per-second billing.

This is the most accurate and fair billing model, especially for shorter calls, as you’re only charged exactly for the time you use.

Which is better?

Can I check VoIP balance and consumptions online?

Yes, you can check your balance and consumption online through the VoIP portal provided by us.

Can I register online for the voip account?

Yes, you can! Click here to register for an outbound VoIP account for your auto dialer or manual dialing needs.

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

Yes, you can test our route/VoIP service with a $1 free credit. Kindly sign up for the trial account by clicking here

Can I check call details online?

Yes, you can access your Call Details report online.

What is the validity of the VoIP account?

It’s for life lifetime.

What is the support for recharge and troubleshooting?

You can recharge anytime by contacting us at:

👉 Microsoft Teams: Click to join us on Teams

👉 Live chat on website: Click to chat with one of our live agents

How to activate Predictive dialing or Answering Machine machine detection function?

Campaign Settings

> Go to the Campaign detail view

> Put 8369 in Campaign VDAD extension or in the routing extension field

Carrier configuration for a Userbased & IP based account

🔧 1. Carrier Configuration: IP-Based SIP Account

Used when your VoIP provider authenticates based on your server’s IP address.

Carrier ID: IP_SIP_CARRIER

Carrier Name: IP_SIP_CARRIER

Registration String: ;No registration needed for IP-based

Account Entry:
[IP_SIP_CARRIER]
disallow=all
allow=ulaw
allow=alaw
allow=g729
type=friend
host= X.X.X.X ; Replace with your VoIP provider’s IP
dtmfmode=rfc2833
context=trunkinbound
nat=comedia
canreinvite=no
insecure=port,invite

Globals String: (Leave Blank)

Dialplan Entry:
exten => _14X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _14X.,2,Dial(sip/SIP_CARRIER_TECH_PREFIX${EXTEN:2}@IP_SIP_CARRIER,55,tToRg)
exten => _14X.,3,Hangup()

Set 14 as the Dial and Manual Dial Prefix in the campaign settings

🔐 2. Carrier Configuration: User/Password-Based SIP Account

Used when your provider requires a SIP username and password.

Carrier ID: USERBASED_SIP_CARRIER

Carrier Name: USERBASED_SIP_CARRIER

Registration String: register =>your_sip_user:your_sip_password@sip.provider.com:5060

Account Entry:
[USERBASED_SIP_CARRIER]
type=peer
username=your_sip_user
secret=your_sip_password
host=sip.provider.com ; Replace with your provider’s domain or IP
port=5060
disallow=all
allow=ulaw
allow=alaw
allow=g729
dtmfmode=rfc2833
context=trunkinbound
nat=comedia
canreinvite=no
insecure=port,invite
fromuser=your_sip_user
fromdomain=sip.provider.com

Globals String: (Leave Blank)

Dialplan Entry:
exten => _14X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _14X.,2,Dial(sip/SIP_CARRIER_TECH_PREFIX${EXTEN:2}@IP_SIP_CARRIER,55,tToRg)
exten => _14X.,3,Hangup()

Set 14 as the Dial and Manual Dial Prefix in the campaign settings

At a time how many campaign we can run?

We can run n-number of campaigns, but it all depends on the number of calls/concurrent calls to be made through the server and server specification, i.e, processor and RAM. The more powerful the server, the more calls can be made concurrently.

What is the remote agents – number of lines function?

In VICIdial, the “Number of Lines” field in the Remote Agents setup tells the system how many simultaneous calls it should send to a single remote extension—usually for cases where no live agents are manually logging in via the web panel.

This feature is commonly used in automated campaigns like Voice Broadcast and Press 1 campaigns, where human agents may not be involved in the first part of the interaction.


🔧 How It Works:

When you configure a Remote Agent:

👉 VICIdial will simulate 10 virtual agents all mapped to the same extension (SIP/1234). This means VICIdial can send 10 simultaneous calls to that extension.

What is remote agents & what purpose we use it?

Remote Agents in VICIdial is a feature that allows you to simulate agent logins and route calls to a single remote extension (like a SIP or IAX2 phone, IVR, or even another system), without needing a human to log in via the VICIdial web interface.

It’s mainly used to:

How Do Agents Log In to VICIdial?

✅ Agent Login Steps in VICIdial


🖥️ Step 1: Open the VICIdial Agent Login Page

(e.g., http://192.168.1.100/agc/ or your domain name)

👤 Step 2: Enter Agent Credentials

User: Agent username (e.g., 1001)
Password: Agent password
Phone Login: Phone extension login (e.g., 1001)
Phone Password: Phone password
Campaign: Select the campaign to log into

➡️ These credentials are created by the admin in Admin → Users and Admin → Phones.

📞 Step 3: Answer the Call on Your Softphone

⚠️ If you don’t answer or the phone is misconfigured, you won’t receive leads.

🎧 Step 4: Agent Screen Loads
Once connected:

What web browsers do you recommend?

Mozilla Firefox and Google Chrome are highly recommended.

What phones will work with 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 VICIdial.

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, it has been renamed 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 completely open-source dialer system.

Common Agent Login Errors & Fixes

👉Issue:“No one is in your session.”
👉Fix: The phone extension did not answer the call. Check the SIP config.

👉Issue: Login freezes at loading…
👉Fix: Your browser may be blocked by a pop-up blocker. Disable it.

👉Issue: Calls are not coming in
👉Fix: Make sure you’re logged into an active campaign and not paused.

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

Check the following settings.

👉Navigate to Users > Choose the user > check if your phone login password matches the Phone extension login password.

👉Phone extension login password can be found by navigating to Admin Settings > Phones

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 reconfigured 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 VICIdial server (to your SIP gateway) should either be GSM or 729. 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.

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 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