Detailed technical information about VoIP2Go

VoIP2Go Customers are welcome to BYOD (bring your own device) and use any make or model of equipment from any manufacturer. Cisco are one of the most popuar brands and here are the minimum settings required to get your device working with VoIP2Go:

Proxy: VoIP2Go Domain Name
User ID: VoIP2Go User ID
Password: VoIP2Go password
Register Expires: 180
Use DNS SRV: Yes
DNS SRV Auto Prefix: Yes
SIP T1: 1
RTP Packet Size: 0.02

ITSP Profile X > General > Name:
VoIP2Go User ID ProxyProxy: VoIP2Go Domain Name
AuthUserName: VoIP2Go User ID
AuthPassword: VoIP2Go password
X_ProxyServerRedundancy - Enable
X_DnsSrvAutoPrefix - Enable
Auto Firmware Update > Method: Disabled
ITSP Provisioning > Method: Disabled
OBiTALK Provisioning > Method: Disabled
OBiTALK Service > Enable: Unchecked
Physical Interfaces > PHONE Port >
PrimaryLine: SP1 or Trunk Group 1

Primary SIP Server: VoIP2Go Domain Name
SIP User ID: VoIP2Go User ID
Authenticate Password: VoIP2Go password
User ID is phone number: No
SIP Registration: Yes

Perhaps the best way to describe the low cost Cortelco devices are that they are cheap and quirky, but can work perfectly well with a little bit of effort. Before reading what we have to say, it's highly recommended that you read the reviews of others who have purchased the devices on Amazon... Cortelco 8211 Reviews

The Cortelco ATA's are not for everybody. If you are a techie and like a challenge, then the price of $11 for the single line 8211 model is very attractive. At just $15, the two-line 8212 model is also very cheap and it might be worth spending that little bit extra based on the 8212 Reviews

Cortelco are based in Corinth, MS and have been around for over a century. We contacted them directly and as of early 2014 they still had a couple of thousand single line devices and a few hundred 2-line devices in clearance stock. It looks like they are trying to clear them on Amazon. If you decide that you want to use any of the Cortelco devices with our Basic Plan, then we can provide you with clear instructions on how to enter the internal settings.

The physical location of servers is critical for Voice over IP. If you are making calls from New York then many of your voice conversations could benefit tremedously if your VoIP equipment is connected to a server located in New York. This is especially important for making and receiving International calls.

Latency, or lag as it is sometimes referred to, is a problem for many users of Voice over Internet services. It happens when you are a long distance from your VoIP server. It can cause delay in your voice reaching the person you have called and vice versa. It can also cause echo.

Quite simply, if you are in New York and looking for crystal clear voice conversations from a Voice over IP service, then you need look no further than VoIP2Go. We have four modern VoIP servers located in a New York Metro data center to ensure that your calls encounter the absolute minimum amount of latency. If there are ever problems with our servers in New York, then your equipment can automatically switch to our lightning fast servers in Washington, Toronto or Chicago.

We have access to a total of five servers between our two partners in New York. One of our partners has a Miami NAP with routes to Latin American which is easily reachable from NYC.

Due to potential delays in every network, it is always a wise decision to choose VoIP servers that are physically located as close as possible to your VoIP equipment. If you make and receive calls from Los Angeles then your phone conversations will benefit hugely if you use servers housed in a Los Angeles Data Center. This is even more critical if you make or receive lots of International calls.

Call it latency, lag or delay, it is the number one problem for a large number of users of Voice over IP phone services. It can lead to echo and jitter, which ultimately lead to frustration. Latency is caused when your VoIP equipment is located a long distance from your VoIP providers server, so if you live or work in Los Angeles, then you should seriously consider trying our servers hosted by QuadraNet in downtown LA.

Luckily for Californian residents and businesses, the Southern Cross Cable hits US shores in Los Angeles making it an ideal hub for International communications with Australia, New Zealand and the whole of Asia.

QuadraNet, Inc.
530 W 6th Street
Los Angeles, CA

The location of data centers is very important for VoIP service. If you are in Chicago, then your voice conversations could become much clearer if your VoIP device is connected to our VoIP servers located in a Steadfast Data Center in downtown Chicago. Using a local data center is particularly important when making or receiving International phone calls as calls encounter the least amount of latency.

Latency is a problem for many VoIP users and it is caused when your VoIP equipment is a long way from the VoIP providers server. It can cause echo and delay at both ends of your conversations and has the potential to become very annoying. If you live or work in Illinois and expect crystal clear voice conversations from your VoIP provider, then look no further than VoIP2Go. We have four cutting edge Voice over IP servers hosted by Steadfast Networks who operate out of state of the art data centers in downtown Chicago, Illinois (350 E Cermak Rd. Suite 240 and 725 S Wells St. 8th Floor).

Redundancy and Failover are specialties of VoIP2Go and should we experience network issues with the Steadfast servers in Chicago, then your equipment can automatically failover to our equally capable servers in New York, Washington, Toronto or elsewhere.

Not only will connecting to our Montreal based servers give residents of Quebec the best possible call quality, it will keep your carbon footprint low. Our cutting edge VoIP servers are hosted by iWeb who provide us with a highly scalable infrastructure that's 100% powered by electricity from clean, renewable energy sources. iWeb's green data centers are located in Montreal. At only 60 miles from the USA border, the VoIP2Go Phone Service can take full advantage of low latency connections to fiber optic networks in New York, which result in lightening fast speeds and crystal clear voice communications to destinations around the globe.

Latency is one of the biggest problems faced by VoIP providers and it is responsible for introducing delay and echo into voice conversations. This can be reduced significantly and even removed completely by connecting to the closest VoIP server to your equipment. If you live or work in Quebec and rely on clear phone calls, then utilizing the VoIP2Go Montreal servers will be one of the smartest moves you can make.

Crystal clear voice conversations can be achieved when you are located close to your VoIP provider's servers. If you live or work in or around Toronto, then your calls will almost certainly benefit from your proximity to our servers.

Our four Toronto VoIP servers are hosted by Amanah and we use a technology called DNS SRV to automatically switch between them. Amanah owns and operates its data center as a direct tenant of 151 Front Street West in Toronto. It is Canada’s premier carrier hotel and one of only a handful of carrier-neutral sites in North America. The location offers unparalleled access and connectivity to the top ISPs in the country. In the unlikely event that the data center becomes unnavailable, then the use of DNS SRV technology ensures that there is automatic failover to our Montreal and Vancouver data centers.

When you rely on Voice over IP technology for phone service, then the last thing you want is delay and echo affecting your call quality. This can be avoided by connecting to the closest VoIP server possible. Whether you are closer to Toronto, Montreal or Vancouver, VoIP2Go are confident that your calls will be free from delay and echo.

If you are closer to Vancouver than Toronto, then our two VoIP servers in the Astute Hosting data center might help improve your VoIP communications. Astute Hosting are a wholly Canadian-owned private corporation in B.C., Canada, operating out of the City of Vancouver

Through a combination of carefully engineered network infrastructure and top of the line upstream bandwidth providers, Astute provides a true performance-oriented, high availability network. In addition to 10Gb links to premium transit providers, their infrastructure is built with redundancy at every level.

The reason VoIP users should connect to the closest server, is to reduce the risk of introducing latency (delay) into calls. Delay can lead to echo, which in turn can lead to frustration. The two Vancouver servers are the newest of ten servers available to VoIP2Go Customers in Canada and they are ideally positioned for making calls to International destinations across the Pacific.

The use of DNS SRV Redundancy in your VoIP equipment or software ensures that failover to our other servers in Toronto and Montreal is both quick and seamless.

Calling Queues allow you to manage incoming calls and have your customers wait in a queue until someone answers their call. Many calls can be held in a queue on a first in, first out basis.

Only 'members' can answer calls from the queue. There are two types of member - Static and Dynamic. Static members are always 'connected' and can answer calls at any time. Dynamic members need to log in first to let the system know that they are available to answer calls.

The advantage of Static members is that as long as they are registered to the VoIP server, they do not have to log into the queue using the *11 and *12 commands. The disadvantage is that Static members are unable to log out of the queue, so the system assumes they are able to answer calls at all times.

There is an option for Music to be played for your Customer while they wait in the queue. You can also setup a Join Announcement. This is where a preset recording plays when a Customer enters the queue. This is often used by sales departments, so that Customers hear a recording of all our products and services as they wait.

Calls are distributed amongst members handling a queue with one of several Ring Strategies:

Ring all: ring all available member phones until one answers.
Least recent: ring the last member most recently called by this queue
Fewest calls: ring the one with fewest completed calls from this queue
Random: ring each phone in no particular order
Round Robin Memory: Remember where were on the last ring pass

The 'Ring in Use' setting lets you avoid sending a call to a member whose device is currently in use.

Mobile VoIP is still a developing technology and here at VoIP2Go we are doing our best to keep our Customers informed about all the latest gadgets, features and tricks to make your Mobile VoIP experience as enjoyable as possible.

Our policy is that when you open an account with VoIP2Go, you should be able to make and receive calls on as many devices as you like. We want you to use it on the network at work, use it on your Mobile device when you are working out of the office and use it on your phone or Mobile device when you are at home.

There are many pro's and con's for each method of connecting to a VoIP server, but in very simple terms, it's best to use TCP for Mobile VoIP where possible to minimize battery usage if you need to receive calls on your phone.

Cellular network Telco's do not make money if you are able to make and receive VoIP calls from your mobile device and some put blocks in place. It's best to use IAX for Mobile VoIP as it can help overcome some of the restrictions.

G711 is the codec of choice on fast Internet connections but it's best to use GSM or G729 on slower networks like 3G or 4G. There's not much difference between the two of them, so it's best to experiment and decide for yourself.

Once again, there is no right or wrong choice of Mobile SIP App and it's all about experimentation and finding the one that suits you best. VoIP2Go suggest you start with Zoiper and switch to CSIPSimple if you are a little more adventurous.

Even the most simple VoIP problems can be difficult to diagnose. There is some freely available software called Wireshark that is the ideal tool for the job, but it requires the use of a network hub which makes setup non trivial. The next best thing is to download a software program called a Syslog Server. There are many available and some of them are free. VoIP2Go recommend the free Kiwi Syslog Daemon which can be downloaded from

After you have installed Kiwi, start it up and it should listen for UDP packets on port 514 by default. If not, go to the File menu, Setup, Inputs, UDP and click on the box - Listen for UDP Syslog messages. Make sure the UDP Port is set to 514.

Cisco/Linksys Device Setup

Log into the web pages of your Cisco or Linksys VoIP equipment and click the Admin link, followed by the Advanced link. Go to the System tab and enter the private IP address of your PC (add :514 on the end) into the Debug Server field. E.g, if your computer IP address is then you should enter

Still on the System tab, set the Debug Level to 3 or 3+H.

Go to the Line 1 tab, or the Line 2 tab if your problem is with line 2, and set the SIP Debug Option to any value to full or 1-line. If your problem is not related to registration then you should select 1-line excl OPT|NTFY|REG. Click on the "Submit All Changes" button to save your changes and you are ready to start debugging.

Obihai Device Setup

From System Management, select Device Admin. Under the Syslog section, enter the local IP address of your computer. If you don't know how to find the IP address of your PC, then enter

Leave the Port setting at the default of 514 unless you have a specific reason to change it. Leave the default Level setting at 7 as this will give you the most detailed information.

Whichever VoIP equipment or Syslog Server you are using, as soon as you lift the handset on your phone you should start seeing messages like 'Off Hook'. When you make a phone call, detailed debug information will be sent to the grid in the Syslog Server and the data should give you everything you need in order to work out what the problem is.