Author: DrVoIP

Peter S. Buswell (aka DrVoIP) is a rare blend of C-level executive leadership and deep technical expertise in telecommunications and cloud infrastructure. With a career that began at AT&T in 1970, he has founded and led multiple industry-defining companies, including Dextr.Cloud—acquired by AWS Global Partner CloudHesive. Peter has held senior roles at firms like Exxon Enterprises, Intervoice/Brite, and Franklin Telecommunications, and has consistently driven innovation from traditional PBX to Amazon Connect cloud deployments. A hands-on architect with certifications from Cisco (CCNP/CCDP, UCCX/UCCE) and AWS (Solutions Architect, Developer, SysOps), he combines strategic vision with engineering precision. Known as DrVoIP, he continues to bridge the gap between enterprise goals and technical execution in today’s evolving cloud communications landscape.

Deploying the ShoreTel Personal Call Manager through AD Group Policy!

Posted on by DrVoIP

Installing a ShoreTel IPBX solution is a process not an event. I have previously published a book entitled “VoIP System Planning Guide” that can be download from the DrVoIP site. This guide covers the basics for planning and managing a VoIP deployment in general and a ShoreTel solution in particular. The “devil is in the details” however and though the process can be understood, the individual tasks required to complete the process generally prove there is no substitute for hands on experience!

Every installation technician comes to that fork in the road that deals with the deployment of the ShoreTel Personal Call Manager software. Deploying the actual telephone instruments is a pure act of labor, but the Personal Call Manager is an act of commitment! Each desktop in the installation will need to be touched by someone, and I do not consider an installation complete until the Call Managers are deployed and operational. There is a component of this effort that involves interVLAN routing, (e.g. getting from the desktop data network to the phone server), but I am now focused exclusively on the actual installation of the PCM software.

There are three strategies that are generally employed to accomplish this. The first strategy is obviously to visit each desktop with a DVD or Thumb drive and load the software! For the installer this is very labor intensive and requires that the install have administrative desktop privileges or maybe even domain privileges. The second option, is to push the software out to the desktops through and email link set from the ShoreTel Director portal to each ShoreTel user. This is a bit less labor intensive, but it still requires the desktop users to have administrative installation rights to their own desktop computers. Most large IT environments do not grant this privilege to plain vanilla users!

The third option, however, has the most promise as being both labor economical while maintaining network security. We can create and Active Directory Group Policy to push the PCM out to the user and have it installed without user involvement. To do this you will need to create a few objects, modify the organization unit containing the computers and users that will be effected by the new group policy. (Refer to Microsoft Knowledge base article 816102). First you create a Distribution point; the create a Group Policy Object, assign a package and then publish your installation package. This strategy is the preferred implementation practice for deployments of any scale and installation technicians should become familiar with the basics of implementing this solution. We will publish a video on both the blog and the DrVoIP site that will demonstrate this solution.

ShoreTel Enterprice Contact Center Skills Based Routing!

Posted on by DrVoIP

The ShoreTel Enterprise Contact Center provides for “Skills Based Routing” an often confusing and misunderstood feature. It is important to note that this function might not achieve the desired result as it is only effective when there is more than one Agent available to accept a call. Lets take an obvious skill, like a Language requirement and assume that our call center needs both English and French speaking customer service representatives. We can assign a skill level to French, but we might not achieve the desired result. If we want to make sure that French callers get connected to French speaking Customer Service Agents, we might consider some other selection options.

For example we could use an Automated Attendant to select English or French and then route to the correct Agent Group. We might also use an IVR application to do a database dip, if they are an existing client, to determine their language preference. Finally, you might have separate DNIS numbers for each language required. These options will achieve the desired result, where “skills based” routing may not.

If we choose to use “skills base routing” as our best fit strategy, first we need to activate skills based routing. This is a system level option and you will find it as the SKILLS tab in entities under system. Each skill gets a value that indicates the minimum level of skill required to process the call. This means that a fluent native language speaker might get a FRENCH SKILL value of 100%. Another agent might get a 75% rating meaning that they have a language proficiency but equal to that of a native speaker. The “skill’ has a minimum value requirement of 75%.

Agents in turn are assigned two values. The first value defines the ability with respect to the requirement. In this example, someone would have to have between 100-75% as a French speaking skill. The second value would be a “preference” value. In this case the expression “preference” is an indication of how much the Agent likes to work with this skill and not your preference for selecting that Agent. This is a very important factor.

Lets assume we have Agent A and Agent B. The selection process is based on the product of the Agents value and preference subtracted from the required skill. The lower the value is most likely to be selected. Agent A has a 50% skill value and a 100% preference; Agent B has a 75% value and a 75% preference. Remembering that we have set 75% as the minimum skill required to handle this call, the arithmetic works like this: Agent A would be (75%- (50X100/100) = 25. Agent B would be (75% – (75×75/100) = 18.75 and as a result Agent B would be selected to handle this call. Again remember that the best fit is only applied when there is more than one Agent to select from! Thus skill based routing ensures that higher skilled Agents in a Group get calls before lower skilled Agents. Under heavy call volume, the skills have less of an effect, so keep this in your thinking when planning to achieve the desired result.

ShoreTel Contact Center Call Select or Agent Select

Posted on by DrVoIP

Agent Call Select OptionsConfiguring a ShoreTel ECC is better understood if we start from the Agent and work back to the caller. In the ShoreTel Enterpriser Contact Center we first define Agents whom we then assign to Groups which are generally the “destination” of a Services. Services are reached through entry points referred to as an IRN or Internal Routing Numbers. The IRN is connected through the PBX via TAPI to the DNIS dialed by the caller. Services can also have Destinations of Scripts which might be an IVR application or database dip to obtain additional customer or routing information.

Agents have both a Call Answer Strategy and a Call Select Strategy. We can search for an Agent based on which Agent has been idle the longest; Terminal, Circular or Best Skill Fit. This is defined at the Service level as “Agent Search Criteria”. What happens, however, if an Agent is a member of multiple Groups and an opportunity exists to have a called present to that Agent from all the Groups that the Agent is a member of? How do we determine which call from which Group should be presented to the Agent? This is where the “Call Select” strategy kicks in. We can assign a primary rule and a secondary rule. The rule can then define if the Call should be selected by the Longest Wait Time; by the Priority of the Call (set as the result of a previous Script or as assigned by the IRN when the call entered the system); and lastly by the “Best Skill Fit”.

Skills-based routing will be discussed in a later blog, but suffice it to say that the Enterprise Contact Center has a superior feature set at a price point that puts this contact center in its own product space. Knowing how to implemented the Call Center should not be left to OJT personnel! Get an implementation team that knows the difference between a ‘dress rehearsal’ and a ‘take’.

Live Answer or Autoamted Attendant?

Posted on by DrVoIP

The genius mathematician and founder of the area we know as Cybernetics, Norbert Wiener coined the phrase “the human use of human beings”. Writing post WW2, his frame of reference was societies growing concern with automation. The concept should now be understood as “using computers to free humans for more productive work”. So when we encounter an automated attendant or “robot receptionist”, we should remember that the repetitive work of greeting, screening, routing, message retrieval and message acquisition are sometimes best done by a machine. We are often asked when setting up a new phone system if we should use “live operator’ or an “automated attendant”. It has been my experience that quality call handling is best done by a human and I would go with live answer. There is a trade off however and here is how you might manage that trade off.

It is generally recognized, unfortunately, that seven out of ten phone calls to our place of business are not clients! They are friends, family, vendors and generally people who know exactly who they want to speak with. If we could free up our human receptionist to answer client calls, the thinking is that we might make both a more efficient call processing mode and at the same time make a much better “first impression”. For this reason, we encourage clients to setup up a “side door” that is always answered by an Automated Attendant. Publish a phone number to “insiders” that routes them through the automated attendant. Publish the company’s main number only to clients. Hopefully this will free the human receptionist to give quality call handling to the most important callers, your clients as that person is now free of the other 7 out of 10 calls.

Trends – Part 3 The Growth of WiMax

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The key word in wireless is “mobility”. Broadband wire line, WiFi, WiMax and even dial-up are technology enablers that provide a solutions for the increasing need to be “connected”. There are issues with each one: broadband service can be expensive, depending on the provider, and it certainly isn’t available in many rural areas; WiFi has very limited range, again limiting coverage, and dial-up is simply slow and can’t come close to meeting requirements for today’s applications. WiMax has gone through a number of changes and with the introduction of WiMax 4G promises to be a viable solution for PBX connectivity independent of location. Ultimately, a network of connected WiMAX towers will drive the deployment of an 802.20-based Global Area Network (GAN), closely resembling cellular networks, but with far fewer towers required to provide the same coverage. This will allow true ubiquitous access across the country or region, providing bandwidth comparable to cable Internet service, at the very least.

The concept of “fixed mobile convergence” moves form concept to reality with WiMax. The ability to move freely from your office PBX extension to your cellular phone number, completely transparently and seamlessly brings the “mobility” functionality into high relief. Companies like ShoreTel already have location based services that enable a PBX telephone extension user to have calls manipulated based on location. Simply stated, when I am in the office my cell phone is a PBX extension, when I am out of the office my PBX extension is a cell phone number. This is completely transparent and requires no change in the users call handling methods. The network sorts it all out for your you.

In today’s market, dual mode phones are already available. I my self use an Apple IPhone (seach “ShoreTel iPhone” on YouTube.com) for video presentation which is a dual mode phone. In the office the iPhone links automatically with our in-house network using an 802.11g WAP. I can retrieve my email, for example, through the wireless access point. When I am outside the office, I can sync with my email using the AT&T cellular network. In the office I have a SIP softphone running on my iPhone and it becomes my PBX extension. When leaving the office, using the ShoreTel “office anywhere” functionality, my iPhone cellular number becomes my PBX extension. This technology will mature with the growing acceptance and availability of WiFI in general and WiMax in particular. As a result, PBX applications will become hardware independent and provide feature functionality that is geographically and device independent.

Trends – Part 2 the impact of SIP

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Lets take a look at the impact of SIP – Aide from the pure technology play, SIP represents a fundamental change in the economics of the telecommunications market. The carriage of telecommunications has been in transition with a steady migration for distance sensitive to usage sensitive pricing. Historically there where three components of the cost of a telephone call: origination, interexchange (e.g. Inter-LATA) and termination. The US telecommunications market has been moving toward a consolidation of service providers. Local Exchange Carriers (LEC) and competitive local exchange carrier (CLEC) is becoming as consolidated and the Inter-Exchange (IXC) carriers. Where do we draw the line on Enhanced Service Providers?

Generally, throughout the rest of the planet, telecommunications services are still owned and operated by government monopolies. Rural telecommunications for much of the planet is predicated on the payment of termination fees. If your favorite telephone company wants to interconnect with your parents in another country, they must pay a termination fee to the phone company in that country. This is not unlike the model of a US based LEC paying the IXC who paid a fee to terminate your phone call in another LEC. A complex price model and tariff structure exists even with the current “bucket of minutes” concepts borrowed from the wireless carriers.

At issue here is the impact of SIP phone calls made through the internet, both public and private. With the growing acceptance of SIP trunking and the development of E164, internet alternative carriage is also pressuring the move from TDM to VoIP. The Electronic Numbering Mapping System (ENUM) provides users with, what marketing people would call “experiential compatibility”. Being able to dial a phone numbers in the manner that users have become comfortable is absolutely essential to the success of the migration to and the adoption of VoIP solutions. SIP and ENUM work together to accomplish this.

The impact on the economics of telephony services is dramatic, both at the infrastructure level and and at the usage level. The cost of building packet switch networks, like the cost of build out wireless networks requires significantly less capital investment. The cost to the users will certainly not be distance based; but access and bandwidth based. SIP also provides for the increased use of multimedia communications solutions, also a bandwidth intensive application.

Trends in the PBX equipment market – Part 1

Posted on by DrVoIP

From time to time a new technology comes along that causes a dramatic paradigm shift with significant economic impact. As it relates to the customer premise telecommunications world, there are a number of trends that have the potential of dramatically altering the telecom equipment market place.   Many industry executives have witnessed the switching equipment transition from electro-mechanical technology to solid state and stored program control switching. Clearly the migration from circuit switch to packet switched technology is reaching Tsunami proportions.  So what happens next?  There are three trends coming into high relief on the technology adoption radar screen  that will have significant impact on the telecommunications marketplace, the economics of the marketplace and ultimately define the employment requirements in that market place.  I would summarize and simplify these trends as the commoditization of PBX hardware; the increased acceptance of SIP and the explosion of high speed wireless technology.  The next few blogs will briefly examine each of these trends at the 100,000 foot level.

This blog will take a look at the commoditization of PBX hardware.  Unlike circuit switched technology, packet switched communications demands that the enormous economic investment necessary to the creation of high speed digital signal processing and the supporting semi-conductor technology be rewarded with high volume production.  Production is a factor of demand and to increase demand, prices must be continually reduced.  This reduction in price is achieved in large part by the adoption of the core technology by a wider base of equipment designers.   This is an economic over simplification for purposes of blogging, but  you do not need an MBA in economics to understand why the  Apple Mac is now built on Intel chip technology.   Nor do you need an MBA in marketing to understand the challenge of trying to convince an increasingly savvy consumer market that your brand of laptop is better than the other guys laptop brand!

As goes the PC hardware market, so goes the media gateway market.  Demand for gateway interoperability will make it increasingly more difficult to differentiate one box of DSP’s from another.   For this reason telephone system manufactures will have to make a fundamental decision:  are we in the the hardware business or are we in the software business?  Are we planning to become the low cost, high performance provider of the VoIP building blocks?  Or are we focusing on the applications (read software) that drive the need for building blocks?   It is my assertion that you can not do both and be successful.   A free market place will demand that “PBX” hardware be separated from “PBX” software.   

Consequently, the skill sets of VARs will be tested yet again.   With the transition form TDM to VoIP, traditional PBX vendors had to develop network engineering expertise.  In the near future, before many VARS have made this first transition, they will be asked to make yet another transition.  Application level software integration expertise will become the underlying skill set demanded of the more successful VAR.   So are you in the business of who can sell a ShoreTel SGT1 cheaper than CDW? Or are you in the business of executing the delivery of underlying technology and application solutions?  Both the supply chain and the distribution channel will once again be redefined and that only question remaining is: just how fast will this transition occur?

   

ShoreTel Star Codes!

Posted on by DrVoIP

starcodes

Though I know we have readers out there who have never seen a rotary dial telephone (or a 33 RPM record), many readers will be familiar with “star codes”. Star Codes have been generally well known to most phone users since the inception of DTMF dialing. Who has not done the old *82 to block caller ID? In the ShoreTel system, basically, every feature has a star code making it possible to use a simple single line telephone set to do advanced features. If you do not have a list of these codes, send support@drvoip.com a request and we will send them right along. The characteristic of star codes that has me excited is how ShoreTel uses start codes with its “office anywhere” functionality. Many systems have the ability to forward a call to your cell phone (pun intended), but generally, the call dead ends there in your hand. How wonderful it would be to be on your cell phone and be able to transfer the caller to another team mate BACK IN THE OFFICE PBX! ShoreTel has a number of start codes that enable you to do jus that. In fact, **+Extension number+## will do just that! There are also star codes for conference e, hold, and access to the other star codes we talked about earlier. Now that it some powerful call processing, would you not agree?

ShoreTel Tools for Checking remote dial tone?

Posted on by DrVoIP

Sometimes there is nothing like a telephone lines mans test set to hear exactly what is going on! VoIP solutions have a double challenge: First it is hard to put a butt set on an IP connection; and secondly what happens when that connection hits a gateway at a remote site? This is always a fun situation. ShoreTel has a “trunk test tool” that has some usability. You can actually bring up the test tool, and right click a particular telephone line and place a call. If you are really in the fast reader group, you can set the test tool options to point the call at your own remote handset. In this way, you are sitting in NY, dialing out on a SF analog telephone line.

ShoreTel has another debug tool that is very useful for “hearing” what is happening on a remote analog telephone line at a site that you just cant get a handle on. You can literally telnet into the switch and setup up a recording session and capture the analog line sounds to a file back at your location. This will enable you to listen to the error message or lack of dial tone, locally enabling you to make the right decision about how to handle the trouble ticket.

The audio output of the analog trunk port is saved on the HQ or DVM server that controls the switch. To do this, follow this procedure:

From the start menu, navigate to the Control Panel> Administrative Tools and locate the IIS Manager;
Right-click on the IIS manager and select Properties. Then enable the ability to write to the FTP server by selecting the Write checkbox and clicking OK. This enable the ability to write to: C:\Inetpub\ftproot
At the command prompt, run the following VXWorks commands: Record2File2 (1, 60, “test” ) Audio data from running this command is stored in the file test_rx.pcm and file test_tx.pcm in the C:\Inetpub\ftp which will be stored as 8000Hz 16bit, Mono and can be listed to using a standard audio application.

Give this a shot the next time you are trying to “hear” if an analog line out at some remote location has any dial tone!

VoIP and SRST/AES Encryption!

Posted on by DrVoIP

Encryption of VoIP traffic was, for some of us a humorous concept. I remembered as a young development professional how much fun it was to use a packet sniffer to capture the bosses packets and reassemble his email over the LAN. Years before that when I worked at the phone company as a central office test engineer, it was not uncommon to find an interesting phone call and plug it into the over head paging system to provide entertainment for the late night test crew. There are times I still think the concept of encryption on VoIP is humorous, but it is becoming less funny all the time as we move toward end to end VoIP with no TDM at all in a world populated by terrorists and other evil doers. In any VoIP environment today, you can at some point use the usual tapping tools to capture a phone call as it hits the TDM gateway and is converted from VoIP to traditional analog or digital signals. From an induction coil to a line mans butt set, you can still intercept a VoIP call as it crosses the TDM boundary.


Now that VoIP is being used end to end, we do need to have a mechanism for encrypting at least the media stream. Today we generally do that with SRTP and IETF standard in combination with AES. AES or the Advanced Encryption Standard was adopted by the US Government and comprises three block ciphers: AES 128, AES 192 and AES256. Each AES cipher has a 128 bit block size with key sizes of 128, 192,and 256 respectively. This standard has generally replaced the former Data Encryption Standard or DES. It is important to understand the difference between encryption and authentication. Determining that a signal is “authentic” and originated from a source we believe to be authentic, and encrypting the contents of that communication are two very different issues. Media authentication and encryption ensures that the media streams between authenticated devices (i.e. we have validated the devices and identifies at each end) are secure and that only the intended device receives and reads the data. We need to encrypt both the media (i.e. the voice) and the signaling information (i.e. the DTMF). In most VoIP systems today, SRTO or secure RTO is implemented to assure media encryption. Understand that this encryption is not passed through to the TDM network, so once the media stream leaves the VoIP environment it is subject to eavesdropping.

Clearly as we are now able to employ VoIP end to end, SRST/AES encryption has very powerful ramifications for both the good guys and the bad guys!