Jul 10, 2015 | SIP TRUNKING
You can continue to utilize a PRI trunk while running a VOIP server and utilizing the SIP protocol internally. However there are several issues that come up due to this configuration.
First, you will need a PRI box, that basically takes the PRI stream and converts it to digital for the server to understand. The hardware aside, this can become difficult to troubleshoot issues with calls from external sources as well as calls originating inside the system to external numbers. While the system will work internal extension to internal extension, external out bound/inbound calls may not, and can result in a line that is simply silent, presents a busy tone, or even rings and then disconnects.
The end result is that your server is working, but we need to know why calls into the system from outside are not, and why the system can not call out of the network.
- Are we actually connected to the internet? We can login to the server via an ssh session or even the web GUI and see if we can hit an external web page, or ping tests, as well as running trace routes. So, if we are successful, this still does not answer why we are not able to make or receive external calls
- We can check our Asterisk CLI and see if we can see the invite on an inbound call. If we do not, and we can successfully reach the internet, then we have an issue along the PRI.
- At this point we would need to login to the PRI box, verify it can see our server and that all it’s settings are correct, and reboot the device
Rebooting of the PRI box will generally solve several connectivity issues, but the question is why? The simple answer is generally the provider of the PRI handoff has made some configuration changes and usually do not notify end users that there will be changes, and this requires the PRI box to reboot to connect back to the PRI trunk.
Another common issue along the PRI line, is the handoffs card will go out, and a resulting call to the PRI provider generally results in the provider on the phone telling you they do not see an issue in the circuit, and will require a few hours of prodding to have a provider technician come to the site and change the faulty hand off card. At that point you will now need to login to the PRI box, and reboot it to bring the phone system back online and functional once again.
When we utilize a SIP trunk, this is a direct internet connection and is already being transported digitally and is ready for the server to accept, with no other hardware conversions required. Your only issues that can arise in this situation would be a failed internet connection, or an issue with the SIP trunk provider directly. However the Sip Trunk providers use several fail over connections so this will be unlikely, and a configuration change from your ISP will not affect your SIP service.
While the overall call quality will not show a difference, the troubles that can arise as well as the added hardware costs and extra configurations of a PRI box that are needed make a direct SIP trunk more economical in the end.
Jun 12, 2015 | Business SIP Trunking, SIP TRUNKING
Are you ready to join the VoIP SIP Trunk trend & take your business ahead?
Switching to SIP Trunk for your telephone needs is the best decision you have made. So how do you know your business is ready to implement SIP Trunking?
Today implementing Sip trunk is not as hard as most think. There is very little hardware required to implement the systems, and if you already have an internet connection, your most likely there.
The major component you will add are the handsets, and when utilizing a hosted solution, you may find a provider that can simply rent you the handsets as well, although that is an additional cost. However they should also be able to provide updated and new handsets over a given time line as you retain a contract with them as well.
Even if you are choosing to host your own VOIP server, the start up is relatively easy as it simply requires a server, a switch, and the handsets. Equipment you would own, and would be reflected in the lower cost of your yearly sip trunking costs.
With a reliable internet connection, sip trunking will be very easy and cost effective and given the added flexibility of utilizing IVR menus for your callers, specialized ring groups, time conditions, as well as conference rooms and video chat.
The added features to sip trunking and the low cost of running it over time compared to initial start up costs, or the switch from the old standby PBX will far better suit business today than to not make the switch.
If you are already running your own domain within your business, the ability to do a self hosted solution will be just about as easy as a hosted solution. Your typical VOIP server will sit outside your network, or if you have one, it can sit in your DMZ, either way your infrastructure will most likely be in place already and this is simply an addition of a server and switch.
If you are not hosting your own servers within the confines of your business, this will still be a very easy switch. Simply adding a server and possibly a switch will do the trick. Location of those devices would typically be near the internet providers hand off, or in the MDF Closet. With the current platforms available to handle the sip trunks you can easily pick up server with a small enough foot print to fit directly into a patch panel as opposed to a dedicated rack. Typically these types of servers come bare bones, meaning there is no operating system installed, but are more than capable of handling medium size business systems in excess of 50 handsets.
While cell service could be considered as an alternative, business will not reap the benefits of conference rooms, complex IVR trees, time conditions, and a host of other valuable assets provided with sip trunking. Whether the solution is hosted off site, or hosted by the business itself, sip trunking is a definite asset to any business that requires phone service.
Jun 9, 2015 | SIP TRUNKING
What is the right SIP trunk provider?
It really all depends on your needs and your understanding of what SIP trunking is?
Several internet providers can provide sip trunking, but not all sip trunk providers can provide internet.
One of the biggest worries you will have with sip trunking is latency, and most carriers can provide that data within their QoS (Quality of Service) reports. Generally (as always) we like to see as small of an amount of packet loss in the transmissions as we can. For voice calls 1% or less packet loss would be an excellent quality call.
You will see codecs mentioned by providers, and if your using an open source VOIP server, it will support all codecs, and this should not be an issue as most carriers conform to a standard that is now well supported throughout the VOIP industry.
If you are not hosting your own solution, you obviously should be concerned with a usable control panel, as well as Call detail records interface so you can see what calls where sent/received and when.This is something you will want to track closely as the number of calls made from your system can affect billing on a monthly basis. The times calls are placed can also indicate a potential intrusion into the system as well.
Packaging can vary greatly from provider to provider as well, the number of channels supported to the cost per call. Careful investigation into pricing packages based on your specific needs of the system your running will make a big difference in your pocket book down the road. Also with VOIP, you are now utilizing E911 which will incur fees when a 911 call is placed, so you also want to be aware that even an accidental call placed can incur the fee, as well as alarm systems that may end up tied into the phone system.
Your SIP trunk may connect back to multiple servers in different locations based on the providers network, and this ensures maximum uptime for your system. It is a simply redundant system set up to make sure you are always connected, so it is a good idea to ask, how that will work based on a given package you may be looking at. After all, if your SIP trunk provider is not connected, you are not connected, so call quality and price may be great, but a provider without any type of fail over plan, will not be as reliable as the provider with a viable plan in place.
Does the provider also have a block of sequential DID numbers available for you? This may not be an issue for everyone, but if this is a part of how you choose to set the system up, it can become extremely important, and I have seen from experience that sequential DID’s can be extremely helpful to business.
If you have designed your own system, or are simply looking to have your system hosted, understanding how you plan on using that system, what the needs of your business, and employees are in regards to that system will supply more than enough information to choose a quality provider for the best price. The first step is to know your needs and how you will use the system.
Jun 8, 2015 | SIP TRUNKING
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Choose a SIP Trunk provider
There are several SIP providers and pricing will vary with the number of channels you need to support the amount of DIDs you have or will require.
- Purchase or Port your DIDs
While you may already have a number of DIDs already, there are charges associated with porting those numbers over to another provider, as with cell phones. If you are not tied to the current numbers, sometimes it can be just as cost effective to simply purchase new DID’s, which can generally be bought directly with the sip trunk provider. In addition you can usually access a block of DID’s within a specific range that can help to make extension addressing more meaningful as well.
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Choose the server OS you will run
- From Free PBX to Elastix for prebuilt images, these work well, and Free PBX is one of the better supported open source systems. Updates occur on a regular basis, and include most of the needed modules and GUI for the set up and future configurations.
- Or you can choose to install any Linux flavor you desire and add asterisk and other requirements on top of that. While some companies choose to do this, the pre built images are generally pretty close to plug and play, and well supported in the open source community
- If you are choosing a solution supported within the Microsoft environment called Lync, you will more than likely already have support from Microsoft in place to assist with any set up issues you would encounter
- Cisco call center manager or Unified Call manager is also well supported by Cisco, and generally you will already have a support contract in place to assist in the set up and configuration.
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Choose the hardware
- From the server that will run the OS, to the switch that will connect the handsets, and the handsets themselves.
- Generally there is not a huge amount of processing power needed to run an open source VOIP system, and any physical server will do fine. Virtual Machines also work extremely well within the open source operating systems using a minimal VM configuration.
- Switches can be PoE or just regular type switchs, the number of ports will depend on the number of handsets you will need, but typically for most small and medium business a 48 port switch is sufficient. Utilizing PoE will enable handsets to not require an outlet nearby for the power supply, however you will still require the ethernet port is close by.
- Handsets are the life blood of the system to the end user, and are highly configurable through configuration files residing on the server side of the system. From when they ring to the type of ring, the speed dial buttons to the display the user sees on the phone. It is highly recommended to research the handsets as much as possible, while some offer excellent looking displays, speaker quality may lack, or while displays may not be as bright and colorful, the other options like speakers, and storing phonebooks can make all the difference in the world based on how they are used.
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Set up your Ports
- If you are currently running ethernet ports to PC, you will also want to add additional ports for the Voice segment of your system. Or, you can choose to utilize VLANS via your switch, and purchase handset that can pass Data through to a PC. This is really a matter of preference and based on your knowledge of VLANS, I myself prefer to have the VLANS and allow the data to push through the phone as I think it provides a much cleaner look to the office space for the end users.
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Implement the Extension Scheme
- Create an extension scheme that is meaningful to you or your business based on your DID block, or maybe an accounting code. On most phone systems extension start at 100 or 1000, or if you are going to use several conference rooms or ring groups, you may start of in the 2xx or 2xxx range, and hold the 1xx/1xxx for conference rooms, and ring groups. There is also the option of utilizing the DID for the extensions, such that you have a DID block of xxx-xxx-1234 to xxx-xxx-4567, you may start your extensions at 1234, and run to 4567 and enable you to assign each extension a specific DID.
The extension scheme is probably one of the most important steps in the set up, and it is often overlooked. Maybe after the fact you wish you would have purchased a PoE switch so you didn’t need those power supplies on the phones, or that you would have used VLANS instead of trying to run so many ethernet cables under the desk. But in the end, you can clean up the cables with cable ties, you can live with the power supply, but day in and day out, you will be tasked with altering names, ring groups, call forwarding, and IVR options within your system, and a solid and well designed extension layout can make all the difference between enjoying administering the system and ignoring the system.
Jun 4, 2015 | SIP TRUNKING, Uncategorized
What are the major concerns when considering SIP trunking?
- Bandwidth of your internet connection
- Cost
- Flexibility
- Ease of use
- Security
Bandwidth of your internet connection
Sip protocol does not generally use a large amount of bandwidth, and can generally run along with a typical data connection for most mid and small sized businesses. That said, the reliability of the internet connection is key to keeping the voice calls flowing and business running.
For larger businesses, redundant or fail over internet connections are recommended to ensure constant connectivity to your sip trunk provider.
Lack of bandwidth or an unreliable internet connection can cause static in the voice, and in some cases can cause several seconds of silence as if the line has gone dead.
While troubleshooting a VOIP issue between two servers for one nationwide company, we found that the internet connection to one of the servers was so restricted that voice loss and silence happened more than successful calls. To remedy the situation a quick reroute of the sip trunk from one server to the other, and then down a point to point IAX trunk resolved the issue until the client was able to bring in a more reliable internet connection.
Generally we are looking at an average of 170kbps of bandwidth for a typical call, and with today’s internet connections, we should easily handle hundreds of calls simultaneously without issue.
Cost
The cost of sip trunking is another major factor, and is generally far more cost effective over PRI and BRI. While the initial cost can sometimes be more, there are several options to successfully utilize sip trunking. You can choose a hosted solution, or you can implement your own.
When choosing hosted solutions, you can expect to pay from as low as $8.95 per month to $29.99 per month on average, and you can choose from several reputable hosting providers. If you choose to host your own, you will simply by DID’s from a provider, and you will supply your own VOIP server, and gain added flexibility in the configuration.
As with many hosted solutions, you will have limited conference rooms, with extras available for a charge, follow me capability can be limited, as well as voicemail to email functions and other numerous features that are available to you when hosting your own server. Choosing to use a hosted solution can result in a lower cost of hardware, in the short term.
On the other side of the coin, while hosting your own server and gaining access to several features available within the system, you will lose the added Call Detail Record features that several providers will supply with your normal monthly service fee. However they can be added to the system, but are not always straight forward to configure or work with, but can provide as much and in some cases much more detail than what a hosted provider may supply.
Choosing to host your own server also requires the hardware, and depending on the phones you may also need POE switches to power the handsets as well.
Flexibility
Sip trunking allows more flexibility than utilizing PRI’s and BRI’s as you can run much more than simply voice through SIP.
Several hosted providers will also have options for video conferencing, chat capability, as well as texting.
Hosting your own VoIP server will also allow the same functionality as hosted solutions.
With a move to offices through out state and countries and even around the world, video conferencing as well as conference rooms are becoming more important, and the flexibility to include these features into one system with a centralized control allows a lower overhead cost and added productivity between sites.
Ease of use
The ease of using a SIP trunking is as simple as picking up the phone and dialing a number, or simply picking up when the handset rings. But what about the back-end side?
If you are using a hosted solution, you can be sure that there will be support for larger issues, and you will have the option of being able to administer several of the smaller easier configurations yourself.
Hosting your own solution, can be tricky, but once you are set up, and you know you can send and receive calls, configurations for the most part are done via a GUI interface and are generally fairly easy and straight forward.
While certain configurations such as an IVR tree and it’s recordings can become large and complex rather quickly, a simple flow chart can help to speed the process and ease the configuration on the server side greatly.
Security
Security is not just a concern for email, or passwords, but for any device connected via the internet today. And make no mistake that most ‘hackers’ begin with VoIP servers.
If you are a small or medium size business then you may want to look into hosted solutions simply based on security. However, with the several available VOIP server operating systems available, their reliability and support, security in regards to the SIP protocol has and is becoming better and better everyday.
Via the use of simple base operating system tools such as IPTABLES, and restricted access via ssh you can greatly increase the security of your system. When those are expanded to include things like Fail2ban and other features within the VOIP applications, systems can be built to be near hack proof.
In my current position, when I first started I rebuilt VOIP severs on a weekly basis as new hackers logged in and erased databases and configuration files, through the use of a few simple and generally accepted admin practices, I have virtually stopped unwanted entries into the servers. Now, that is not to say that there are no longer attempts, because the fact is that programs exist like sipvicious that will scan IP ranges looking for traffic across port 5060, and then it’s as simple as running something like Kali Linux to get into the system. But implementing the above mentioned system, you can easily see incoming attempts long before access is gained, and simply block those address.
You can continue to utilize a PRI trunk while running a VOIP server and utilizing the SIP protocol internally. However there are several issues that come up due to this configuration.
