If you ask your average IT professional what a T span is, the usual response will be that it is a 1.5MB connection to the internet. Ask your average telecom tech what a T span is and you will be told it is 24 channels of dial tone! As a VoIP Engineer what a T span is and you should get the answer:, “what do you want it to be”? One of the great challenges of implementing a VoIP solution is the absolute requirement that the implementation team possess an interdisciplinary skill set. The solution demands expertise in a range of specialized skills including IP network, switching, routing, supplementary telephony services , server technology management and application call flow integration. If the user group is going to fully realize the benefits of a VoIP implementation, then each of these specialized areas of technology are going to be necessary to a successful deployment. Traditional telephony vendors are comfortable with all things TDM. They like to punch things down on 66 blocks and use “butt sets” to test for “dial tone”. Network professionals have their area of comfort as to Microsoft or Linux server professionals. Call Center professionals understand caller greeting, salutation, screening, call routing, message acquisition and message retrieval at the application level, but seldom understand the underlying technology. At the end of the day, you can shop the internet and find out who can sell you a shiny new telephone thing cheaper, but finding a team that can execute the delivery of a VoIP solution is worthy of the time you would invest selecting a new CFO! You need to work with a team that can demonstrate proficiency in each of the required discipline and accept responsibility for every aspect of the implementation. From concept to “go live”, the voip solution provider you select must know the difference between a “dress rehearsal” and a “take”.
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.
760 Area Code Change to 442 – San Diego
760 AREA CODE
Who Will be Affected? All customers with a 760 number will have to change the way they dial. The new 442 area code will serve new customers in the same geographic region as the current 760 area code, which extends from Bridgeport in the north, south to the Mexican border, Camp Pendleton on the west, and east to the state line.
What Will be the New Dialing Procedure? To complete calls from a landline phone, the new dialing procedure requires callers to dial 1 + area code + telephone number. This means that all calls in the 760 area code that are currently dialed with seven digits will need to be dialed using 1+ area code + telephone number.
When Will the Change Begin? Beginning May 2, 2009 … you should begin using the new dialing procedure whenever you place a call from the 760 area code. If you forget and use the old dialing procedure of dialing just seven digits, your call will still be completed.
Beginning October 24, 2009 … you must use the new dialing procedure for all calls. If you do not use the new dialing procedure, your call will not be completed, and a recording will instruct you to hang up and dial again.
Beginning November 21, 2009… new telephone lines or services may be assigned numbers with the 442 area code.
What Will You Need to Do? In addition to changing your dialing procedure, all services, automatic dialing equipment, or other types of equipment that are programmed with a 7-digit telephone number will need to be reprogrammed to use the new dialing procedure. Some examples are life safety systems, fax machines, Internet dial-up numbers, alarm and security systems, gates, speed dialers, mobile phone contact lists, call forwarding settings, voicemail services, and similar functions. Be sure to check your business stationery, advertising materials, personal checks, and your personal or pet ID tags to ensure the area code is included.
What Will Remain the Same?
· Your telephone number, including current area code, will not change.
· What is a local call now will remain a local call regardless of the number of digits dialed.
· The price of a call, coverage area, or other rates and services will not change due to the overlay.
· You can still dial just three digits to reach 911, as well as 211, 311, 411, 511, 611, 711 and 811.
Who May You Contact with Questions?
contact support@drvoip.com for assistance in programming any changes to your ShoreTel phone system!
What is a ShoreTel DVM and why do I need one?
What exactly is the value of a Distributed Voice Mail Server (e.g. DVM)? What are the pro’s and con’s of installing one? Does it have any impact on resiliency (not redundancy) as it relates to business continuity in the event of server failures? ShoreTel has a distributed architecture but like all other VoIP solutions there is only one “read/write” database and that is a component of the ShoreTel architecture aptly named the HQ server. IF this server goes down and the R/W database is unavailable configuration changes can not be made throughout the “single image” installation.
Installing a DVM at the same level, or in the same site as the HQ server, provides a high degree of resiliency at comparatively low cost. At the HQ site, put all your HQ users on a DVM. If the DVM goes down, the HQ will pick up the heavy lifting for the Users on the DVM. If the HQ goes down, the DVM users will still have VM and AA services. As of today, there are three services, however, that are NOT distributed in the ShoreTel architecture. These services are known as Workgroups, Route Points; and Account codes. If you lose the HQ server, you will lose these services for all sites, even if they have a DVM installed at that site!
As it relates to low cost business continuity options, we like to install a DVM at the HQ site, but we want all switches at all sites to be managed by the HQ server. This usually provokes a heady discussion, but here is our reasoning. The real value of a DVM is to keep VM and AA media streams off the very expensive WAN connections. Remember that a DVM can fail up, which means the HQ server can take over Voice Mail and AA processing for the users at a site that has a failed DVM. It makes sense to put the users at a remote site on the DVM at that site, but does it really make sense to have the switches at that site managed by the DVM at that site?
We think not. Lets separate the issue of Users and Voice Mail from issues like TAPI, Workgroups and Personal Call Managers. We need to remember that if a server goes down, the switches managed by that server will lose all the TAPI information for the phones that it controls. This means you will have no functioning Workgroup Agents and not ability to monitor those Agents. Additionally, the Personal Call Managers will not work for any extensions on switches managed by the down server.
Given that Workgroups is not a distributed service, if the HQ server goes down, you will not have Workgroups anyway. If the DVM at a remote site goes down, the HQ server will proxy for that sites Voice Mail and Automated Attendants. Given that the HQ server was managing the switches at that remote site, you will not lose any of the PCM functionality highlighted above. It occurs to us that this is a better place to be. Let the HQ manage all switches and use the DVM’s for Voice Mail services for the users at remote sites! Use a DVM at HQ for additional resiliency.
Voip Network Monitoring
We have been actively working with VoIP since 1999! Since 2001 we have installed well over 10,000 ShoreTel desktops and one characteristic of these VoIP environments has surfaced into high relief on the radar screen here in technical support: A VoIP solution is only as good as the computer Network it runs on! Network Monitoring – a Necessary Evil? When someone mentions network monitoring, most network administrators immediately start thinking: overpriced, large server requirements, difficult to install, time-consuming to configure. If those hurdles are overcome, then there’s a potential rainbow at the end of the road: Immediate notification of problems, faster problem resolution, less downtime of services. That equates to happier & more productive users, and a more profitable organization. What’s interesting to realize is that the vast majority of companies all want to know the same things with their network:
- When do problems happen?
- Where are the problems?
- Why do these problems exist?
We have decided to create a product that eliminates all of the hurdles and answer these same questions no matter how large or complex a network was deployed.
We can now:
- Deploy and auto-discovers your entire network in just a few minutes
- Continuously monitors the health of every device and interface on your network
This allows for some proactive analysis that includes:
- Quickly learn which interfaces in your entire network are discarding packets
- Perform a call path mapping of the health of every interface used in a VoIP call
- Run a call simulation from any computer to any IP endpoint (including router interfaces)
- Know what your current Internet utilization is – live (updated every 2.5 seconds)
- Learn the switch and port where your VoIP phones are connected
Contact us today and we will send you a FREE completely operating network monitoring system for your evaluation. Send a return email that lists:
- Company Name
- User Name
- User email address
- User phone number
And we will email you the download link and evaluation license code! Our only requirement is that you be a ShoreTel system user.!
SIP Softphone on Asterisk, CISCO and ShoreTel
Configuring SIP extensions has become one of the most requested support questions we receive. Asterisk, CISCO and ShoreTel all support SIP extensions. Candidly, SIP is an excellent protocol and one that most IT professionals are already familiar with. This is an application level protocol and looks an awful lot like HTTP etc. Ever go to a webpage and get a 404 error, “page not available”? Well SIP error messages are very similar. If you get a 404 SIP error message, it means phone not available. Though H323 and MGCP are certainly more mature protocols, they are most useful for “call processing”. SIP has the added dimension of being useful for presence information. Setting up SIP extensions on ShoreTel is a breeze and we have setup up everything from CISCO through Polycom desk phones and a handful of wireless handsets as well. ( I am working on getting my iphone connected to ShoreTel using a wireless SIP phone and I am breathlessly awaiting the release of 3.0). X-Lite makes a great SIP softphone on ShoreTel and I have included a soundless SIP configuration video for those of you familiar with ShoreTel administration. This configuration is built using cally the same in 8.1 so enjoy!Version 9 of ShoreTel, but it is basically the same in 8.1 so enjoy!
Fax on VoIP IPBX systems!
When VoIP gateways first started hitting the market back in 98′, the vendors tried to packetize DTMF and quickly learned it did not work well. For this reason they quickly learned to “regenerate” DTMF at the outbound gateway rather than packetize and truck it across the network. Moving modem tones is even more challenging and most VoIP solutions discourage attempting to send modem tones over an IP connection. If you can get it to work at all, the modems will negotiate down to about teletype speed or about 300 baud. (Now there is a device and an term you don’t hear much about anymore). Given that a fax machine generates modem tones, faxing over an IP connection is equally challenging. ShoreTel makes it possible, for example, to attach a low end fax server as a couple of analog IPBX ports. Incoming fax calls to the IPBX system, can reroute these faxes and even regenerate the DID number as DTMF to the fax server enabling fax to email applications. What we are beginning to see as the SIP market matures, is that the phone companies are bringing PRI circuits to the customer premise disguised as traditional TDM circuits. Your Telco interface may in fact be a ShoreGear T1, but you are interfacing to an Integrated Access Device that is converting SIP signaling to SIP and then on to the Telephone companies softswitch. Currently, this is a big problem for fax machines connected as analog lines to the host IPBX. True Fax over IP is going to require a T38 interface and fax machines and servers that can support this protocol. The message here is, make sure you know what you are connecting to! True TDM or a SIP trunk! Knowing the difference will enable you to properly handle fax traffic.
VoIP and Microsoft Outlook Integration?
Does your Microsoft Outlook Integrate with your phone system? This functionality is getting to be the “minimum daily adult requirement” feature in the VoIP vendor space. We all just expect that our phone system “knows” about our “contacts”. We don’t dial phone numbers anymore! We enter Names and the phones system gets the number out of our contact list and places the call. Often, an incoming phone call to our desktop, will cause our contact information to be displayed. Some integrations enable your phone system to change user profiles and call handling modes based on your Outlook contact. Of late I have been wondering how far this integration can go? I mean, if I have a conference call scheduled in my Microsoft Outlook, shouldn’t the phone system know about that? My thinking is the phone system should just call me and remind me of the conference and then ask me to approve joining the meeting!
ShoreTel compatible audio conference server?
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!
ShoreTel Contact Center – Integrated Agent Tool Bar
The ShoreTel Contact Center provides two strategies for call management at the desktop. We have found that the basic ‘agent tool bar’ is an excellent solution for call center desktops in which different shifts sit at the same desk, use the same computer and phone. It is easy enough to program the tool bar to prompt the agent for there log-in information. In this way, you can set up your phones with extension numbers that multiple agents can use. When an agent reports to work, they go to their assigned desk and log-in, using an extension that might have been used by a different Agent on the previous shift. The ShoreTel Contact Center keeps track of what agent used what extension during what time slot. On desks that are dedicated to an individual agent, the agent tool bar can be integrated into the ShoreTel Personal Call Manager. The Personal Call Manager looks like a Workgroup Agent call manager, but the tool bar indicates “contact center” and the drop down list contains information that is appropriate to the ShoreTel Contact Center or Enterprise Contact Center. There is one additional advantage of the stand alone Agent Tool bar. You can push custom parameters, named “call profiles” in ShoreTel documentation, to the Agent Tool bar. We find that you do not have the same flexibility with the Integrated Tool Bar inside the ShoreTel Personal Call Manager.
Call Center Design
It is very normal to start planning a call center around the concept of DNIS, Groups, Agents and Skills. Experience has taught us, however, that call center design needs to start with a more important concept. Call Centers are performance oriented. We measure the number of calls presented, the number calls answered, the number of calls abandoned, average talk time, the average time a caller is waiting in queue for service and any number of other key parameters. We want to know call disposition, agent availability and how much time was used for lunch and breaks. Is the staff sized appropriately for the volume of calls that are arriving? Do we over flow excess call volume from one group to another group? How often do we do this and how long did callers waiting in the original queue before this happened? Do we Interflow from one call center to another? These parameters are all key characteristics of a call center and the very parameters we want to measure. For this reason, call center design should begin with the actual report that we want call center to produce in order to verify our performance! Lets construct the report we want to use to manage our business and from this document, we can best work backwards to creating the groups, agents, skills and Queue messages.