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Managed Voip

Considerations when Picking a Managed VoIP PBX

Not all things are created equal, and when considering a new phone system for your business, not all Cloud Based Managed VoIP Providers are the same. Before you sign a contract, there can be huge differences among Hosted VoIP Providers.

Features – What features are most important to your business? Does your business need auto attendant, voicemail sent to an email, mobile twinning (sending calls to both a cell phone and desk phone at the same time)? Does the receptionist want to see who is on the phone? How about the ability of having “hot desking”, the ability of logging into anyone’s phone and have it appear as your own. This feature works great for medical practices who have rotating staff working the front desk. Don’t forget to ask office workers what features they could use.

Equipment – What about the brand of phones that are used? Is the equipment proprietary or can it be used with other Managed VoIP services providers? Should you purchase the equipment or rent each handset and what are the advantages? Make sure you are getting quality VoIP phones from a quality manufacture, the last thing you want to happen is finding out the phones you bought are not good quality. Does each user on the system need a fancy phone with lots of features, most employees only use two or three features. Do you really need a cool looking conference room phone or will a basic handset do the trick? Many newer phones have excellent speaker phones, so a basic handset may work fine. A good provider should be able to offer multiple phone options as your business grows and expands.

Pricing – Many providers offer confusing or different pricing options. Some offer unlimited options that may be simple to understand but you pay for features not needed. Another consideration is whether to rent or buy phones. With some customers it makes sense to buy, but what happens when the phone breaks, who is responsible? The cost of renting phones has dropped dramatically, however pricing and features vary greatly. Make sure you understand how the companies long distance calling is priced; contrary to what many believe, Hosted VoIP is not free phone service.

Call Quality – This is where customers get burned and have poor VoIP call quality and get disappointed. It is important to know the difference between BYOY (Bring you own Bandwidth) compared to “managed VoIP” using a private MPLS data network. Some carriers provide an extra layer of call quality by using a managed router. Make sure you know the difference between managed and unmanaged services, there can be a big difference in call quality.

Vendor Experience – This is one of the most important considerations when considering a Managed VoIP phone system. VoIP (Voice Over Internet Protocol) has been around for many years and many service providers are now selling Hosted VoIP via the internet, out of car trunks, basements and garages. It would be disastrous for a business if the phone company went out of business and had control of your phone numbers? Make sure you find out how long the Hosted PBX provider has been in business, how many customers they support and the types of customers.

Disaster Recovery – It is very important to make sure you understand the providers network and how many POP’s (point-of-presence) they own and manage. Does the hosted PBX provider have built in intelligence that can determine when a business’s on-site phones stop working and can they re-rout calls to different numbers? How many network operation centers does the provider have, east and west coast only?

Summary – Managed VoIP PBX offers advanced features previously only available to much larger business all for a great value. Hosted or Cloud PBX phone service compared to traditional solutions offers no-hassle phone service without ongoing maintenance, service contracts, costly hardware and onsite trip charges. While a hosted PBX offers customers ease of management; an onsite or Premise PBX is can still be a more cost efficient solution.

freepbx

Set Up Extensions on a Cloud Based FreePBX

One of the best things about modern VoIP systems is how flexible they are when it comes to how you deploy them. You can use them on an appliance, virtualized, or on a cloud-based service like Amazon AWS, Google Cloud, or Microsoft Azure. Each configuration has a slightly different technique to making everything work, and one of the first challenges is registering extensions. For this post, we’ll focus on the general concepts of setting up extensions for a cloud based (hosted) solution with FreePBX.
If you’ve never heard of FreePBX, and you’re in the market for a new VoIP system, you should start doing a little research ( and also call VoIP Supply). To be brief, it’s a turn-key PBX solution that uses Asterisk, a free SIP based VoIP platform. Sangoma, the makers of FreePBX have created a web user interface for Asterisk to simplify configuration. They’ve also added an entire security architecture, and have added a lot of features above and beyond what pure Asterisk (no user interface) provides, such as Endpoint Manager, which is a way to centrally configure and manage IP Phones.

FreePBX isn’t the only product out there to do this, there’s quite a few out there actually, but FreePBX has really raised the bar in the past few years and has become a very series solution for the enterprise. Don’t let the word “Free” in FreePBX lead you to think it’s a cheaply created system.

 

FIRST, A LITTLE ABOUT VOIP CLOUD SECURITY:

There’s a huge benefit to hosting a VoIP system in the cloud, you have to deal with very little NAT. Why is that good? SIP and NAT generally do not cooperate with each other. It’s very common for SIP header information to be incorrect without a device such as a session border controller (SBC), or a SIP application layer gateway (SIP ALG). When deploying a system on premise, you will always need to port forward SIP (UDP 5060) and RTP ( UDP 10,000-20,000) at a minimum. Also, you’ll need to make sure these ports are open on your firewall. This helps direct SIP traffic to your phone system, similarly as if you had a web or mail server.

Of course, there are security concerns when exposing SIP directly to the internet, and the same concerns apply for a hosted system, but when dealing with a cloud solution, you are generally given a 1:1 (one to one) NAT from your external IP address to the VoIP system’s internal IP. A 1:1 NAT ensures all traffic is sent to the system without any additional rules. Some cloud services place an external IP address directly on your server, increasing simplicity.

If you’re reading this, and are becoming increasingly concerned, you’re not wrong. If you’re in the technology field, you’ve probably been taught that exposing any server directly to the internet is wrong, bad, horrible, and stupid. Generally speaking, that’s all correct, but luckily many cloud service providers will offer the ability to create access control lists to place in front of your server, like the one below from Microsoft Azure.

Cloud service Microsoft Azure

This gives you the ability to control access to specified ports, source, and destination IP addresses. Additionally, FreePBX has built in intrusion detection (Fail2Ban), and a responsive firewall, allowing you to further restrict access to ports and services. Is this hack proof? No, of course not. Nothing is hack proof, but I have run my personal FreePBX, exposed directly to the internet, with zero successful attacks. No, that’s not a challenge, and you can’t have my IP address. You can, however, have some of the would-be hacker’s IP’s (see below).

freepbx hackers ip

 

 

If you’d like to learn about the firewall that FreePBX has put together, go here. I’m not suggesting, that this is just as good as placing an on-prem VoIP system behind a hardware firewall, but the results so far are that it works very well. Using a cloud solution will always be at your own risk, so do plenty of testing and take whatever measures needed to secure your system (disclaimer).

 

SETTING UP (REMOTE) EXTENSIONS:

One of my favorite feature of a cloud based system is that all extensions are essentially remote extensions. This means you can place a phone anywhere in the world, in theory, with an internet connection, and place calls as if you were sitting in the office, or at home. There are some variables to this configuration, mainly restrictions on whatever network your phone is connected to, but generally speaking, it’s a useful and user-friendly solution. Now, for the rest of the article, I will assume that you know how to create an extension on FreePBX and have basic familiarity.

The first thing I typically do when deploying a new VoIP system is to define all of the network information for SIP. This is important for both cloud systems, and on-prem, Specifically, you need to tell FreePBX what networks are local, and which are not. To accomplish this, proceed to Settings > Asterisk SIP Settings, and define your external address, and local networks.

General-SIP-setting

 

 

Next, if you have your firewall turned on and you should make sure SIP is accessible. You’ll notice in the below image that the “Other” zone is selected, meaning I have defined specific networks that are allowed under Zones> Networks. To allow all SIP traffic, you can select “External,” but you would be better off enabling the Responsive Firewall, which rate limits all SIP registration attempts and will ban a host if a registration fails a handful of times.

CHAN_SIP

 

Also, something to pay attention to: Make sure you use the right port number. By default, PJSIP is enabled, and in use in FreePBX on port 5060 UDP. I will generally turn off PJSIP and re-assign 5060 USP to Chan SIIP. This can be adjusted under Settings > SIP Settings > Chen SIP Settings, and PJSIP Settings.

Bind-Port

 

Once the ports are re-assigned, you MUST reboot your system, or in the command line, run ‘fwconsole restart.’ I also like to tell FreePBX to use only Chan SIP. To do that, go to Settings > Advanced Settings > SIP Channel Driver = Chan SIP. PJSIP is perfectly funcitonal, but for now, I recommend you stick with CHAN SIP as PJSIP is still underdevelopment.

We should also assign the global device NAT setting to “Yes”. This will be the option used wheneber you create a new extension. Without making this the global default, you will have to make this change manually in each extension, when you’ll likely forget to do, and your remote extension will not register. This setting lets FreePBX know that it can expect the IP phone or endpoint to be external and likely behind a NAT firewall. To change this global setting, go to Settings > Advanced Settings > Device Settings > SIP NAT = Yes.

SIP-Nat

 

Lastly, make sure your extensions are using SIP, if you haven’t turned off PJSIP. You can convert extensions from one channel driver to the other within an extension’s settings.

SIP type

 

At this point, you should be able to register your remote extensions to your cloud based FreePBX system. If you are running into trouble, run through these troubleshooting steps:

  1. Check the firewall – Allowing SIP? Are you being blocked?
  2. Check Fail2Ban (Admin > System Admin > Intrusion Detection) Are you banned?
  3. Check that your networks are properly defined in SIP Settings
  4. Verify you are registering to the proper port
  5. Make sure the extension is using the proper protocol
  6. Debug the registration attempt in the command line – Authentication problem?

I hope this article sheds some light on the topic of cloud based VoIP systems, and how to set up extensions for that system. I also hope this saves you a few hours in troubleshooting if you are not well versed in FreePBX configuration. As a friendly reminder, before you make any changes to your production system, take a backup, or snapshot, and always test your changes.

PBX as an automation tool

Have you ever thought about digging into the underbelly of your phone system to see what it can really do? Chat room integration? CRM/ERP integration? How about extending it to your employees’ cell phones? That’s not even the end of the possibilities. If you can imagine it, it’s probably possible.

Over the years we have implemented many of these integrations and have found that over time they have been valuable to provide rapid information and workflow automation.

Finally, make sure to work with educating your users to champion better tools in the workplace!

Enjoy the tech!

Help for Asterisk AA50 including issues, how to rebuild compact flash filesystem, and workarounds

First, I would like to say that the AA50 is not a recommended product.  Actually, I think it's the opposite of it.  I would recommend an analog Phone with a voicemail recorder before I would recommend one of these things.  Why do I have such harsh feelings towards it?  Well, support personnel is unable to realize that a PBX has major issues if it reboots randomly and prevents you from leaving voicemails or getting voice prompts.  I even tried to make them understand by explaining to them that the problem is not an advance or unsupported feature, but one that's critical to the basic intended functionality of the device itself.  My response was "It's not meant to be used as a full PBX".  Secondly, they told me the issues are being worked on, but they haven't figured it out yet.  Uhh… my support ticket was created about a year ago!  Response "Do you know how hard it is to rewrite a firmware?"  I'm a very patient and understanding person, but if you fail to recognize a critical issue with a product at such a simple level, I feel my point will never be accepted.  Just imagine if Toyota took a year to fix their brake problems or say the cars weren't suppose to be fully used that way…. 

I'm proud to do Digium's job for everyone by providing the public community a work-around and documenting what I've learned.  Hope this help others.  As for the AA50, I will never buy anything solely and directly made by Digium again.  Buy Sangoma and use open-source Asterisk.

Background: http://www.keycruncher.com/blog/2009/11/02/digium-confirms-major-issues-with-aa50-voip-appliance-spotaneous-reboots-and-memory-card-write-lock-a-review/

Symptoms:

  1. The system reboots randomly and frequently
  2. The system loses access to the compaq flash filesystem frequently, thus no voicemails or voicemenu prompts or even backups.
  3. The system prevents you from deleting voicemails due to the issue with Symptom 2.

Detail Description:

Basically, the reasons are:  Memory leak(s) (Symptoms 1) and Memory card write-locks (Symptoms 2,3)

Work-around:

Create an automated cronjob to reboot the system on a nightly basis.

  1. Create a script (reboot-24hrs.sh) in /etc/config (use this directory because it's backed up to the local storage; not flash storage)
    #!/bin/sh
    sleep 86400
    /bin/asterisk -rx
    reboot

Edit /etc/config/rc.local and add /etc/config/reboot-24hrs.sh &

What if you wanted to rebuild your compact flash card?  The answer is simple:

  • The appliance on startup (/etc/rc) mounts the compact flash using this command:  "mount -t ext3 /dev/hda1 /var/lib/asterisk/sounds"
  1. /sbin/create_sounds (Formats the compact flash memory card and creates the proper sounds directory.  It also downloads the files from the Internet)
  2. /sbin/update_tz (Downloads time zone files from the Internet)
  3. /sbin/update_phoneprov (Downloads phone provisioning files from the Internet)

What the buzz is about: Virtualization and Consolidation

I know everyone has been hearing this continuously for the last 3 years or so, but what does it actually mean?  How does it help the IT department and how does that help the business?  While the two words of Virtualization and Consolidation are separate topics, they often go hand in hand.  I believe the reason for this are to take advantage of new hardware capabilities and new virtualization technologies.  Lets say you bought a new server and wanted to consolidate and migrate all the data from your file server and your mail server.  You’ll essentially end up with a server with more utilization.  Sure, you’ll save a little on time and the electric bill, but that isn’t going to give you the “WOW” factor when it comes to analyzing your ROI. You’ll soon realize that consolidation usually will not fully utilize the full capabilities of your new hardware.  Unlike oil and water, implementation of Virtualization creates a symbiotic relationship with Consolidation.  One could also say it’s the catalyst to a higher ROI.

Why is there a push now?

  • 64-bit operating systems allow for a significant increase in memory
  • Multi-core CPUs creates effective use of processing power
  • VT-enabled CPUs support virtualization specific instruction sets which increase the effectiveness of hardware.
  • Microsoft is now in the market, which usually means it’s growing and here to stay.

Summary of benefits of Consolidation and Virtualization:

  • Lower TCO (power and cooling requirements, less physical assets, reduce time and resources need by IT staff, reduce licensing requirements)
  • Increased flexibility (backups, snapshots, migration, quicker provisioning of new servers)
  • Space savings
  • Makes use of the full capabilities of your new hardware
  • You already have the capabilities, you just need to implement it.
  • Fail-overs can be performed practically instantaneously
  • Upgrades to a new server in the future is greatly simplified
  • It’s much easier  to make  the  resulting  infrastructure resilient  for  business  recovery  and  continuity  solutions

When will you know it’s “GO TIME”?

  • You are trying to cut costs
  • You are trying to increase performance
  • You will be purchasing a new server
  • You spend too much time focused on maintenance
  • You are implementing a software “refresh”
  • Your servers have multiplied to where you have a management problem
  • You need IT to work projects that drive the business
  • Your backup solution is inefficient and ineffective
  • You have a need for a development environment

Advantages of Voice over IP (VOIP)

What Is VoIP?
Voice over Internet Protocol, or VoIP, is the current technology that allows people to transmit voice signals through the internet instead of over the phone. Most people have already become acquainted with the idea of sending voice over the internet through the use of headsets or microphones, but only a few realize the unique differences between the two.

While a direct connection to a single person or a separate server allowed users to chat with each other using microphones, users still had to have telephone services in order to receive out of network calls. VoIP eliminates the obvious limitations of in network voice communication, and expands it above and beyond our expectations.

Much like an e-mail, users don’t have to pay to send or receive them. E-mails can go anywhere users have set up a mailbox, at any time. Imagine e-mail transforming from text into voice, and virtual mailboxes becoming phones. The result is a completely free, new form of voice technology capable of sending voice from an internet line and converting it into a signal anyone with a phone can receive.

Cost Advantages:
Businesses, especially smaller or medium sized companies, are always in need of more cost-effective tools and solutions. Businesses invest thousands of dollars in order to save money over a longer period of time. VoIP is a service that can potentially provide the results companies are looking for, in an even shorter amount of time. By instantly cutting costs with fewer drawbacks, VoIP has become a popular solution.

  • Telecommunication systems can be merged with current networks, allowing a business to save on the cost of network infrastructure.
  • Remote Web-based interfacing eliminates the need for on-site representatives to repair or troubleshoot phone network issues.  Costs associated with on-site repairs are practically negated.
  • While saving money, VoIP users have found that the service provides much more than the average office phone services. While there were certain limitations with phone services (such as busy lines and expensive remote location calling bills), VoIP has sought to break these limits. Not only do clients receive phone services for nearly no cost, but they also receive tools tailored to make manage and design the network how they want it to be. VoIP puts the client in control.
  • Single IP networked VoIP lines enable extension dialing to expand to multiple, or even distant locations.
  • Applications are all extended to employees at any corporate location (including temporary or remote locations), including, but not limited to: conferencing, voice mail, unified communications, and click-to-dial services.
  • VoIP telecommunications systems are easily simplified into a single network (combined with data networks), allowing for easy management, and the elimination of multiple networks.
  • Remote troubleshooting and management through web-based interfaces. Settings can be changed for specific employees remotely, and without the need to contact service providers or phone system manufacturers.

The 3 evils of Voice over IP (VOIP)

Many of the world’s largest telephone companies are committed to replacing their existing circuit switched systems with voice over IP systems. These packet switch voice over IP systems allow them to transport a significant portion of their traffic with IP. Surprisingly, many calls made over telephone company equipment are already being transported with IP.
Packet switched voice over IP systems are in principle as efficient as a synchronous circuit switched systems, but only recently have they had the potential to achieve the same level of reliability as the public switched telephone network or proprietary PBX equipment. With the invention and implementation of RTP (real time protocol) and SIP (session initiation protocol,) voice over IP has the technological base to obsolete the circuit switched public switched telephone network.

– BY Paul Mahle
Asterisk and IP Telephony / Paul Mahle
Copyright 2003, 2004 by Signate, LLC.

VoIP provides enhanced teleconferencing and remote teleworking to maximize internal productivity, save money and simplify management.

So, you are interested in implementing a VoIP system for your small business, but are unsure of the capabilities of your network. It can be broken down into 3 steps:

1.Determine how well the network is running

2.Deploy the voice over IP service

3.Verify that the service levels are working correctly.

How do you know if you current network is up to the task? What criterias determine if your network is Voice-enabled capable? What are the optimimum factors in running a smooth and clear voice over IP system?

The 3 evils of Voice over IP networks.

1.Delay (minimum of 150ms, use Cisco RTPC + LFI)

This is the time it takes voice to travel from one point to another on the network. It can be measured in one direction or for the entire round trip. The calculations of delay usually involves Network Time Protocol (NTP) and clock synchronization and reference clocks.

2.Jitter (the optimal jitter buffer should fit the network’s differential delay, Cisco’s LFI)

This is the variation in delay over time from point to point. The higher the variation, the more degraded the call quality will be. The amount of tolerable jitter on the network is affected by the depth of jitter buffer on the network equipment in the voice path. When more jitter buffer is available, the network is more able to reduce the negative effects of a broad variation. Unfortunately, a buffer can also be too big. This would increase the overall gap between packets.

3.Packet Loss (less than 2.5-5%, use QoS that differentiates between data and voice packets.)

Packet loss refers to the packets of data that are dropped by the network to manage congestion. Data applications are very tolerant to packet loss, as they are generally not time sensitive and can retransmit the packets that were dropped. Dropped packets in a VoIP network appear as noise in the conversation and may require the speaker to repeat or retype the last word or sentence, which is clearly undesirable.