VOIP Tech Blog

I would like to kill  call center challenges with one blog!   In many call center environments, it is possible that an agent has a Direct in Dial (e.g. DID) number that a client might call and by pass the entire call center process!   For a call center manager, this is very frustrating!  Your create a Contact Center to organize the flow of calls to your Technical Assistance Center, and clients by pass the process by calling your Agents/technicians directly!

Clearly you can eliminate this by not giving DID numbers to Agents, but we end up doing this to facilitate an orderly problem resolution strategy.  You might give a client a “home work” assignment and ask them to call you back.   They might object to being put at the back of the queue again, so you give them a DID number that gets them directly back to you.  The challenge is, that this call would not be accounted for in your contact center reporting!   So how then do you provide this feature and also create a mechanism for enabling your contact center to capture all calls for Agent Performance reporting?  One answer is to establish a “service” that has only one Agent in it!   Then build out a DNIS/DID route point to IRN relationship that brings that DID number into the call center and routes it to the Agent.

Actually, it is not a bad strategy!   The Contact Center can now account for all calls, even the ones that reach your Agent through a DID number and  you can apply normal Contact Center routing tools like “over flow”, which brings me to my second point!   Is it possible to overflow calls to more than one other group?  Based on more than one over flow time in queue?

The Contact Center provides a wide variety of methods for handling callers in queue awaiting service.   One of the more interesting concepts is the ability to “over flow” a caller from one group to another group based on the amount of time they have been waiting in queue.   This contact center parameter is set within the “service” and can be found in within the tab labeled “Overflow”.

In figure One below, you can clearly see that we have a number of services defined, including a service named TAC1.   Let us assume that this is a technical service group and that TAC1 is comprised of Agents/technicians that include Agent Gandalf, Kipling, Regan and Jack.      You can also see that a service has been enabled  by the name “Direct Kipling”.    This service was created to enable callers to Kipling’s DID number to be brought into the Contact Center, and routed to Kipling even though they did not enter through the TAC1 service.   Hopefully, we can now count the phone calls Agent Kipling is handling that otherwise would not be reportable!

In Figure One  we can also see that we have set an “Overflow” counter that,  should anyone be in the Kipling Queue for 10 seconds they will over flow to the TAC1 queue.  What is interesting is that when you overflow in the ShoreTel Contact Center,  you don’t leave the queue you are in, you basically add another queue containing agents that have a cumulative effect on the call.    Should that “overflow” not result in an available agent and the caller continues to wait for service, you can actually set a second timer that would “overflow” (e.g. expand the number of agents ) to yet another queue.

In Figure two, you can see the original call in the Queue for Kipling.  After the pre-configured overflow interval is met, the call is distributed to the “overflow” queue but is also available to the original queue.  You can see this in Figure Three, by noting that the call is now in queue in both the Kipling and TAC1 queue.   In this way if an agent becomes available, in either the original queue or the “overflow” queue, the call will be answered.   Had we set up a second interval timer in the Service, we could expand the number of Agents to include the additional group specified by that timer.  This is one of the more interesting, if not misunderstood capabilities of the ShoreTel “overflow” concept!

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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.

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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.

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In a VoIP environment the WAN circuit is generally engineered to handle X phone calls of a specific codec. For example you might plan out a circuit that supports 10 simultaneous phone calls across the WAN between sites. You select the G711 codec and plan each phone call at 82KPS per call. This would require that there be a minimum of 820KB of bandwidth available or approximately 55% of a full T span. Given that the WAN connection also supports data applications, we want to assure that Voice does not take all available bandwidth! Interestingly when people complain about the bad quality of a VoIP call, it is generally the result of exceeding a bandwidth limitation, If you engineered the circuit for 10 calls, when the 11th call is placed, not just that phone call is trashed, but all 11 phone calls are destroyed! For this reason VoIP systems in general and ShoreTel in particular have strategies for limiting the number of calls across the WAN. In ShoreTel there is a parameter entitled “Admission Control Bandwidth” located in the Sites definition in the ShorewareDirector administrative web portal. This parameter assures that a call will not be set up between this site and another site, if that phone call would exceed the bandwidth setting. This generally eliminates the 11th phone call on a circuit designed for 10 simultaneous phone calls! ShoreTel switches or media gateways, know about the bandwidth they would consume when setting up a phone call and can take action based on this ACB parameter. We need to apply solid WAN engineering practices to the circuit planning however, as the ShoreTel switches will not know if that bandwidth is actually available! So it is possible that the ABC parameter will allow a call to be setup, but bandwidth may not actually be available as other data applications might be consuming more than planned bandwidth at that point in time. For this reason, we need to prioritize voice and data with queuing strategies in our WAN routers, the subject of yet another blog!

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Has the economy hit VoIP? My thinking is that it has and in a very positive way for both integrators and voip service providers. Throughout my career I have always been amazed at the number of traditional key telephone systems sold into the small business segment of the telephone equipment market place. If you added up the numbers as published by the publically reporting companies in this segment, extrapolated an average SBE equipment size requirement you would conclude that every man, woman and child in this country already owns a telephone system! So where do all the new SBE phone system come from? Having said that, this number is getting harder to calculate as the companies that you could track in this segment (e.g.. Vodavi, Comdial, etc) have morphed into something else and generally as a direct result of the increasing acceptance of VoIP in that market segment (e.g. Covad, Packet8, Vonage et. Al). The economy even when it contracts, causes companies that survive to change their shape and size. We have witnessed a growing demand for companies seeking to spin off a branch office as a standalone business. If you have a CISCO Call Manager, for example, and you were running SRST at the branch office, we are servicing requests to convert the branch to a standalone CCME. Business Partnerships dissolve, the partners create new entities and split a perfectly good ShoreTel IPBX in two! Service providers in general, seem to be showing increases in VoIP revenue segments, specifically in the SIP environment. Cbeyond (CBEY) seems to be trading at near its 52 week high. Generally, my thinking is that VoIP is a good place to be no matter which way the economy moves!

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Prior to version 7 of  ShoreTel, backing up your ShoreTel system was very straight forward. There was a single folder in the root directory named d:\ Shoreline data. This folder contained all the information that was required to completely restore your ShoreTel system from a bare metal server in the event of a major disaster. The folder contained the configuration database, which at the time was kept in Microsoft Access. It also contained all of your recorded prompts for Automated Attendant, your voice mail messages, all of your Call Detail Records and softswitch related information. You could easily identify this one folder and make it a part of your normal system backup process for your company. With the introduction of Version 7 of ShoreTel the company began to migrate away from the Microsoft Access database and move toward the MySQL database. First they moved the Call Detail Records and with Version 8, the entire configuration database had migrated to MySQL. For this reason the database backup process for a ShoreTel system has changed. The process must now include the backup of two MySQL databases and the aforementioned Shoreline data folder. ShoreTel does provide a few BAT file examples of how you might do this, but if you want to automate the process complete with a schedule you will want to consider using some other tools. We recommend the use of SQLyog and include a copy on every server that we support or install (just another reason to have DrVoIP do your ShoreTel maintenance). Send an email request to drvoip@drvoip.com and we will send you a tech note that details this process or you can watch this silent video linked below!

How to Backup a ShoreTel IPBX Version 7+

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SIP trunks can be used for a wide variety of call processing  solutions, some of which are very cost effective as in Free!   We have used  SIP tunks to interconnect ShoreTel systems over the Internet, complete with four digit dialing between systems!    Most recently,  we have been able to create a ShoreTel compatible audio conference server at a fraction of the cost normally associated with these servers.   The most costly part of  this conference solution  was the ShoreTel SIP Trunk Licenses at $50 per port and the Server Hardware.    SIP is a wonderful solution and enables you to interconnect a variety of exciting options to your VoIP solution that are vendor specific yet provide the highest level of interoperability.  We now provide an audio conference server, (you provide the server) free to all of our ShoreTel annual support subscribers.  We will send you a link to download the server, just for letting us quote your support contract renewal!

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