TOTLCOM offers our customers the opportunity to pay their bills two simple ways.
1. By Mail:
Attn: Accounts Receivable
65 Hangar Way
Watsonville, CA 94026
2. By Phone:
Contact one of our Customer Care Specialists at 1-800-300-5500.
These acronyms refer to specific types of charges on your bill. MRC stands for Monthly Recurring Charge, and typically refers to the monthly payments you make for your services. NRC means Non-Recurring Charge, and is the money you pay initially for installation and initialization services.
A DID cost recovery fee is a government-mandated fee charged by all telecommunications providers to cover the cost of 911 service and various charges incurred by the provider. It is assessed at the same rate for each DID.
If you need a copy of your invoice, simply call one of our Customer Care Specialists at (800) 300-5500 or Contact Us for assistance.
Late fees are assessed to the billing cycle two months prior to the current billing cycle. Therefore, if your payment is late, the late fee will not appear until another billing cycle has passed.
Your invoice charge includes your Monthly Recurring Charge (MRC) as well as any additional taxes, late fees or additional charges that may be assessed.
Each business is unique in its communications requirements. Your bandwidth needs depend on how many users you have at your location, and what those users are doing. TOTLCOM offers premium fractional T1, T1, fractional T3, T3, and optical data connections. Our Account Executives can help you determine which of these options is best for your situation.
For most customers in most geographic locations, it takes about 35-40 days to install a new T1 connection. TOTLCOM coordinates every step of the process, from signing the contract to scheduling the installation. Additionally, our customer support team is ready to assist you 24 hours a day, seven days a week.
Our T1, NxT1 and T3 (or DS3) data lines are private connections with private ports. We believe our small and medium-sized business customers deserve the speed and security that can only be delivered by a dedicated, private line. Our pricing includes the port and the loop charge together, providing for a seamless, guaranteed data connection.
Yes. TOTLCOM believes consistent, reliable data and voice connections are essential to success for our small and medium-sized business customers. We pride ourselves on providing this service, and this guarantee is tied to our industry-leading SLA. You will always receive the specified speed capability.
TOTLCOM realizes that when your internet access is down, your business is not making money. Most of the problems are such that we can solve them remotely. We have extensive relationships with each carrier, so if there is an issue, we have the ability to resolve it quickly and efficiently. When you sign with TOTLCOM, you receive the added value of our experienced, dedicated in-house service and support geared for small and medium-sized business.
This depends on your network. If you want to add voice or other SIP trunking, that would determine the IP addresses you need. This depends on the size of the company. Each device that is connected to the Internet through your router requires its own IP address. TOTLCOM’s experienced Sales Engineers will guide you through the process of determining your needs and provisioning the required resources.
TOTLCOM offers small and medium-sized business a number of options in terms of equipment choice. An Account Executive will guide you through the process, helping you to decide whether it makes sense to purchase or lease your equipment, and what equipment to purchase or lease, based on your business plan. Call 800-300-5500 today to find out how TOTLCOM can help your company succeed.
TOTLCOM will handle connecting the data to your business location. All you need to provide is a router and cables. TOTLCOM’s experienced Account Executives will provide the correct equipment during the initial order process, making it as seamless as possible, so you can get back to business.
The process of making and receiving voice transmissions over any IP network. IP networks include the Internet, office LANs, and private data networks between corporate offices. The main advantage of VoIP is that users can connect from anywhere and make phone calls without incurring typical analog telephone charges, such as for long-distance calls.
Many businesses can save money by switching to business VoIP for their voice communication needs. Our Account Executives can help you explore options and find out how much your business can save. Additionally, the option of running voice and data communications over the same circuits is another way to save money.
Definitely. Depending on your communications needs, VoIP has the potential to deliver significant savings on communication, letting you allocate valuable resources where they should be allocated: growing your business. Learn how TOTLCOM’s business VoIP solutions can give your company the competitive edge.
Many current phone technologies are capable of running VoIP. If you currently have a PBX, our Account Executives can help you determine whether you’re ready for VoIP. TOTLCOM also offers hosted IP-PBX, allowing you to avoid the hassle of maintaining your own equipment. See how TOTLCOM’s Hosted IP-PBX service can streamline your operations.
SIP is a process that allows for creating, modifying, and terminating multimedia sessions with one or more participants. These sessions include Internet telephone calls and multimedia conferences. SIP is widely used as a protocol for Voice over Internet Protocol, or VoIP.
SIP trunking is a means of leveraging the powerful SIP technology to provide dedicated multimedia support to a physical location. A SIP trunk is a connection between your IP network and the SIP network of your telephony provider – in essence, a SIP trunk is a phone line. TOTLCOM is the leader in SIP trunking technology, with an active and experienced research and development team and 24×7 customer support.
SIP easy to use, is scalable, and saves companies money. With SIP, you can buy exactly the number of lines you need. You can route calls to any domestic physical location, and you can leverage your existing infrastructure to maximize your investment potential.
Internet Protocol Private Branch Exchange. IP-PBX is communication equipment located on the physical premises of the business. It is the backbone of an office communications system.
TOTLCOM offers small- and medium-sized businesses the functionality they expect from a traditional hosted VoIP solution at a fraction of the cost. VoIPZone provides your business with a broad array of voice features that simplify your communications. Visit our VoIPZone page or contact one of our Account Executives – they can help you determine if VoIPZone is suited to your situation.
TOTLCOM contracts with the leading national carriers to offer you the best, most reliable internet access, at the best price, to support your voice needs. TOTLCOM is a complete business communications solutions provider. We offer inbound, outbound, and two-way calling with a variety of available features. Contact your Account Executive for specific details on how TOTLCOM’s VoIP service can give your business an edge over the competition.
E911 is an improvement to the traditional 911 Emergency phone system. E911 provides emergency operators with a physical address for each incoming call. This offers obvious public safety advantages. Some VoIP providers (like Vonage) direct 911 calls to a central operator, who will gather the necessary address information and forward it to the authorities. E911-compliant systems connect you – and your address – directly to the local public safety officials. Our ELS phone lines are fully E911-compliant, offering you and your employees complete peace of mind.
TOTLCOM provides superior SIP trunks and call-routing technology. Our routing networks are structured to provide the most direct route for your calls. TOTLCOM employs multiple call gateways, assuring faster and better connections for your calls. This means reduced latency and coverage of over 90% of the domestic service area. All of this translates into superior business performance. Contact one of our Account Executives to learn the benefits of TOTLCOM Business VoIP.
It depends on the type of PBX you currently have. If you have recently made a large investment in phone technology, it might make sense for you to go with our SIP trunking service, which would allow you to leverage your current investment with the latest technologies. If you favor a more hands-off approach, our feature-rich Hosted IP-PBX service might be just what you need. Our Account Executives can assist you in making this determination.
TOTLCOM provides local numbers in more than 5,300 domestic service areas. At this time, we do not provide international numbers.
TOTLCOM maintains a stock of more than 2 million voice-capable SIP trunks, offering full Local Number Portability (LNP) compliance. We support local call origination from 5,300+ rate centers, covering 300+ markets. Our service footprint covers 93% of the U.S. population, with 1,000 to 10,000 phone numbers available in each unique rate center. We can request specific numbers, but cannot guarantee them. Our number porting service allows you to move the custom numbers you currently have to a TOTLCOM VoIP system.
Probably. Along with many other features, TOTLCOM supports number porting from almost all service providers. We can configure your system with your current numbers, offering your customers and employees a seamless transition.
Typically, we can get your new Business VoIP system up and running in 3-5 business days. We maintain a large stock of local numbers in more than 300 markets, so we probably have the numbers you need in stock. If we need to secure additional numbers, the process will take 5-10 business days.
TOTLCOM offers local numbers in more than 5,300 rate centers in the U.S., covering 300+ markets and serving more than 93% of the U.S. population.
In many cases, our customers can use their current telecommunications equipment. If you currently have an IP PBX or SIP-ready equipment, you will need no additional hardware. We can also help you acquire the equipment necessary to convert your analog voice equipment to VoIP technology.
The amount of bandwidth for a call depends on the compression method (or codec) that is being used. The g.711 codec requires about 80-90k per call, while the g.729a requires about 30k.
TOTLCOM supports both the g.711 codec and the g.729a codec.
TOTLCOM is a complete SIP trunking communications system provider. At this time, we do not support IAX trunking.
TOTLCOM recommends a T1 data connection for the best call quality. T1 lines offer the best combination of uptime, stability, and reduced latency. Our Account Executives can help your business experience the increased productivity of a T1 line and the efficiency of VoIP communications with our bundled service packages.
TOTLCOM offers one, two, and three-year VoIP and data contracting. By signing a longer contract, you can lock in lower rates and reduce some installation charges. Additionally, TOTLCOM offers the ability to re-negotiate pricing options at the end of a contract term.
TOTLCOM offers some of the most comprehensive SLAs in the industry. Our U.S.-based customer support team is available 24×7 to solve all of your communications issues. Read our customer testimonials and hear what our customers have to say about the support they receive from TOTLCOM.
TOTLCOM has the industry’s leading VoIP SLA. Our VoIP service is guaranteed to have maximum uptime and minimal latency and jitter issues.
TOTLCOM knows that your small and medium-sized business needs maximum flexibility when it comes to scaling your operations. You can add a single DID line at any time during your contract for the same low initial rate. Unlike some of our competitors, we will not force you to buy packages of lines you don’t need.
TOTLCOM’s Business VoIP service includes one DID per SIP trunk ordered. Additional DIDs may be purchased in packages at competitive rates.
TOTLCOM offers industry-standard pricing of $0.029 / minute for toll-free lines.
TOTLCOM offers very competitive international rates. Contact one of our Account Executives for specific pricing details.
TOTLCOM can transfer your existing phone numbers to your new Business VoIP system. It typically takes 3-4 weeks to complete the porting process, depending on how many numbers you have to transfer. We realize that speed is essential; most of our small and medium-sized business customers have found that the process can be completed in about 30 days.
Absolutely. Flexibility is a hallmark of TOTLCOM’s small and medium-sized business VoIP service. We can add additional DIDs at any time. Leave us your contact information so one of our Account Executives can contact you.
E.164 Numbering is the international standard for assigning telephone numbers. TOTLCOM DIDs are E.164-compliant, which means that phone systems anywhere in the world can reach them.
If you are having difficulty with your TOTLCOM services, you can get the ball rolling quickly by opening a trouble ticket online. Point your browser to the TOTLCOM website, then click on Email Customer Support. Additionally, you can call one of our Customer Care specialists at 1-800-300-5500. The specialists are trained to get you the answers you need. To help speed up the trouble ticket process, please make sure to include the following information:
When you sign a contract with TOTLCOM, an installation representative will guide you through every step of the process, from scheduling to initialization. We acquire the connection to the local and national carriers, activate the service, and coordinate service personnel. Additionally, we will provide you with a detailed chain of command, so you know who is responsible for every step of the process.
TOTLCOM organizes the entire installation process. The customer has to be on-site during the activation process and have their router, as well as the correct cable connection from the T1 to the router. TOTLCOM’s experienced Account Executives will provide the correct equipment during the initial order process, making it as seamless as possible, so you can get back to the business of doing business.
TOTLCOM works directly with the carrier to make the installation process as quick and painless as possible. For most small and medium-sized business customers, the process will take 30-45 days, depending on the location of the installation. Our experienced service and installation team will guide you through the process, every step of the way.
Yes. In some cases, if the customer’s site is ready, the process can move more quickly. Ask one of our Service Representatives what you can do to make sure the process is as short as possible.
TOTLCOM recognizes that small and medium-sized businesses often need their data connections to be on line as soon as possible. With some carriers, the installation process can proceed more quickly with an expedite order. This feature costs more up front, but results in a shorter installation process. With an expedite, the carrier will have an install done in 15-20 business days instead of the regular 30-40
A demarc is the door to your T1. The carrier will make a connection from its network to your physical location. It looks like a normal phone jack, and is where you “plug in” to the internet.
A demarc is the main point of entry to the facility, which is determined by the carrier. That’s where everything comes into the building. With a demarc extension, the carrier will extend the demarc from the point of entry to the customer’s specified location.
This cost varies with the carrier. Some carriers do it by the hour, others have a pre-set fee. It could also depend on whether the T1 is already installed, or if it is requested before the install. The carrier will add more fees if the line is already installed, so be sure to specify where you want to “plug in” before your installation is scheduled.
No. If customers want an extension, they need to have it set up beforehand.
MPOE stands for main point of entry. This is the same as the demarc.
The carrier determines the MPOE. The customer does not have input in this process. This is the point of having a demarc extension.
A smart jack is the same thing as a demarc. It is where you “plug in” to the Internet.
Installation of a new T1 line typically takes 30-45 days. A TOTLCOM installation specialist will guide you every step of the way. TOTLCOM values transparency and accountability throughout the process.
TOTLCOM works hard to get your business up and running as quickly as possible. Our installation specialists will get your order provisioned in the same business day.
TOTLCOM believes in complete transparency throughout the ordering and installation processes. Our engineers will provide detailed notes and due dates for the various tasks so that you can keep abreast of the progress. Occasionally, a date may have to be moved back to accommodate a carrier’s schedule. We do our best to minimize any such extensions and will always let you know what is going on.
TOTLCOM works to get you up and running as soon as possible. Our billing department will begin billing two days after we inform you that your system is ready for activation.
TOTLCOM takes customer support seriously. We employ highly trained engineers to keep you up and running. Many problems can be resolved remotely, while some problems require a more in-depth approach. Our SLA details the specific customer service guarantees you can count on.
TOTLCOM’s Customer Care Specialists work hard to make your company’s move as seamless as possible. They will make the necessary arrangements with your carrier to set up your service and get it connected at your new location.
If at any time you choose to terminate your service with TOTLCOM, let us know 40 days prior to the final date on which you need access to your service. We’ll handle the rest.
To adjust the contrast of your IP 530/560 phone, press and hold the ‘Mute’ button as you repeatedly press the UP or DOWN arrows on the ‘Scroll’ button next to the phone display.
The ShoreGear-120, ShoreGear-60, and ShoreGear-40 voice switches send a heartbeat to their associated IP phones once a minute. If the heartbeat is not acknowledged within approximately four seconds, the switch considers the IP phone to be offline or unavailable. The switch continues to broadcast the heartbeat every minute. Any currently offline IP phone that returns an acknowledgement is considered online and available.
Extension Port or Universal Port in Extension Mode:
On hook voltage: -47V DC.
Off hook loop current limit: 28 mA.
Ringing Voltage: 47VAC RMS.
Ringing Frequency: 20 Hz.
Ringing waveform: AC Trapezoidal on both Tip and Ring, no DC offset.
Ringing capability: 3 REN (Ringer Equivalence Number).
Ringing Cadence: Under program control.
Loop resistance: 200 Ohms maximum.
Impedance: 600 Ohms.
Universal Port in DID Trunk Mode:
Idle voltage: -47V DC.
Off hook loop current limit: 28 milliAmperes.
Loop resistance: 1500 Ohms maximum.
Impedance: 600 Ohms.
DID Type: Wink Start.
Universal Port in Loop Start Trunk Mode:
Off hook operational loop current required: 10 mA minimum, 120 mA maximum.
Off hook voltage, at 20 mA loop current: 7.5VDC maximum.
Ring detect threshold: Must detect above 16.5V AC RMS, must not detect below 13.5VAC RMS.
Ring Frequency Range Detected: 14.71Hz to 83.3Hz.
Ring Cadence Detected: Under program control. Minimum 680ms on, followed by minimum 2000ms off. At these minimums, it could take four rings to answer. With US ringing (2 sec on, 4 sec off), it takes two rings to answer.
Ringer load: 0.0B REN (Ringer Equivalence Number).
There are two ways to reboot the IP 100 phone:
The IP phone reboots whenever power is reapplied. You can accomplish this by disconnecting the power cable momentarily and then reconnecting it.
You can also reboot the IP phone by pressing four keys simultaneously. The four keys are:
Hold down these keys until the display shows that the IP phone is rebooting.
Yes, it is compatible with all versions of ShoreTel including 6.1
The power consumption is as follows: See Attached Data Sheet for additional details.
|Model||PoE Power rating||Idle||Active|
|IP 565g||Class 3||4.2 W||6.9 W|
|IP 560g||Class 3||4.1 W||7.1 W|
|IP 560||Class 2||3.4 W||6.4 W|
|IP 265||Class 2||3.5 W||5.9 W|
|IP 230g||Class 2||4.0 W||5.9 W|
|IP 230||Class 2||2.9 W||4.4 W|
|IP 212k||Class 2||3.1 W||5.1 W|
|IP 115||Class 2||2.6 W||3.7 W|
|IP 110||Class 2||2.8 W||4.1 W|
|IP 8000||Class 3||4.3 W||8.2 W|
You are trying to bring new phones on line in a new installation. You are not sure which configuration switch the phone is using. You are wondering if you can direct the phone to a specific switch.
In a DHCP environment, when an IP phone is booted, it receives an IP address from DHCP and the required FTP Server information. Once the phone reaches the ShoreWare Server the phone should communicate with a Configuration Switch. If the phone cannot reach one of the configuration switches, the phone will not be added to the system. Configuration Switches are added to HQ by default once the first two switches are added server. You must manually change the Configuration Switches if you would like to assign specific switches. The IP addresses of these switches are downloaded to the IP phones whenever the IP phones are booted. If you have configured the IP phones to boot without a DHCP server, you must set the IP address of the configuration switch manually (MGC Address). These switches communicate with the ShoreTel server to determine which switch manages calls for a particular IP phone.
The ShorePhone IP 230 and 230g includes a 24 character x 5-line display, 120 x 35 pixels, however it does not have a backlight.
The 655, 565G, 560G, 265, 212k. 115, 110, and BB24 do.
The IP 565g phone supports the Bluetooth V2.0 standard, which is backwards compatible with Bluetooth 1.x headsets. Compatible headsets need to support the hands-free or headset profile, as defined in the Bluetooth standard.
Even though internal testing of third-party headsets for the IP 565g has been performed, ShoreTel does not certify or support products from headset vendors.
Because of the inherent environmental inconsistencies in the locations in which the IP 565g may be deployed, there is no one best headset solution that is optimal for all environments. ShoreTel recommends that customers test the headsets that work best in their environment before deploying a large number of units in their network.
The primary reason that support of a Bluetooth headset would be inappropriate for an installation is the potential for undesired noise, such as crackling or popping sounds.
Such noise may be heard by the remote party or by both the remote party and the IP 565g user. Causes for the noise can include:
In some instances, the mechanics or electronics of various headsets can cause remote parties to hear their own voices echo back when they speak to IP 565g Bluetooth users. Sometimes this can be mitigated by turning down the volume setting of the phone and/or headset.
Testing performed in the ShoreTel labs has shown that the Plantronics 510 and Plantronics 655 have performed well with the IP 565g, but it remains the customer’s responsibility to test in their environment.
Please contact TOTLCOM for the latest list of additional supported headset models.
Prior to ShoreTel 7, ShoreTel IP phones have four different sets of ring tones, with each set consisting of one tone for internal calls and another for external calls. The user was free to select from the four tones as desired. However, in some densely-populated work environments, four tones were not enough for users to be able to distinguish the sound of their phone from that of their neighbors’ phones.
Starting in ShoreTel 7, Administrators have the ability to load custom ring tones on the IP phones so that each user can have a unique ring tone.
One set consisting of two custom ring tones (one for internal calls and one for external calls) can be loaded onto each IP phone by the ShoreTel Administrator. This new set of tones will displace one of the existing sets of ShoreTel ring tones.
Ring tones must be in Microsoft Wave File format (.wav). These custom ring tones are not provided by ShoreTel, but numerous web sites offer files that can be downloaded for free. A quick search should yield a large number of possibilities. Once a set of custom ring tones have been identified, the system administrator must load these ring tones onto a user’s IP phone via an FTP server.
Please refer to ShoreTel 7 Release Notes for details. This feature is supported on all ShoreTel IP phones except the IP 100 or IP 8000.
Call Manager supports Microsoft SP2 and SP3.
XP service pack 3 (SP3) has been certified with ShoreTel 8 (build 13.9.9403.0 or higher) and ShoreTel 7.5 (build 12.15.3601.0 or higher).
To edit the DNIS Digit Map, click Edit DNIS Map from the appropriate Trunk Group edit page to invoke the DNIS Digit Map.
For each DNIS entry, type a meaningful name into the Dialed Number field. This can be a mix of digits and alphabetic characters. This friendly description will be presented in the Call Manager as well as in the CDR Reports.
Enter the digits you expect from the service provider when this number is dialed in the Received Digits field.
Select a Destination extension using the Search button and then click Add this record.
An Operator Call Manager user must have the proper permissions to have the “Extension Monitor” option available.
When account code collection is enabled or required for a user group, calls placed via the telephone or the Call Manager are routed to the account code extension. The Account Codes Service prompts the user to enter an account code followed by the â#â key. If the account code entered does not match the digits in a stored account code, an explanation message is played and the user can enter an account code again. When a matching account code is collected, the call is placed according to the originally dialed number.
The call permissions define which dialed numbers will be directed to the Account Codes Service for user groups configured with account codes. For calls that are redirected to the account codes extension, the call will be completed with the trunk access and call permissions of the Account Codes Service.
This structure imposes two sets of permissions to outbound calls:
Yes, users have the option to setup Call Handling Mode Delegation. This feature is helpful for users who may want their personal assistants to change their call handling mode for them.
To delegate call handling:
Step 1 – From the Personal Options page of the Individual User, click Delegation (which is found to the right of the Current Call Handling Mode). The Call Handling Mode Delegation page appears.
Step 2 – From the left column, select the user(s) to whom call handling will be delegated and click Add.
Step 3 – To remove a user from call handling delegation, from the right column, select the user you want to remove and click Remove.
Step 4 – The User who wants to change the call handling mode would then use their “OPERATOR CLIENT or WORKGROUP SUPERVISOR CLIENT” and right click on the users call handling mode to delegate and change.
IP phones and other IP endpoints communicate with ShoreGear switches via MGCP, a device control protocol. The relationship between the switch (call manager) and the phone (gateway) follows a masterâslave model. MGCP, an industry-standard protocol, is used to:
SELF-AUDITED LICENSES: These are licenses that MUST be purchased from ShoreTel – but are not tied to a key. This means that the customer simply goes into the ShoreWare Director and physically enters the number of licenses of a specific type that they have purchased. A license is counted as being “required” as individual ShoreWare system users are configured to use that type of specialized call manager – or as additional servers are added.
You will find information regarding the Softphone within Personal Call Manager under Help, Contents and Index, Softphone.
From Call Manager (ShoreTel 7.5 and Below) – You can place the calls from the SoftPhone by using the Call Manager interfaces just as you would with an analog or IP Phone. The Soft Phone must first be taken off-hook (Simply Press the TALK Button) for a call to be placed however. If a call is placed from Call Manager while the Soft Phone is on-hook you will hear some ringing to alert you that the Soft Phone is still on-hook. Also Handsfree Mode can be used to suppress dial tone on the Soft Phone just as with other phones.
On the PC you want Call Manager installed on, open an IE browser and enter the URL as follows:
The ShoreWare Client Install page appears. After reviewing the information on this page, click the Install button
The InstallShield Wizard downloads the installation files (showing the progress of the download), “unpacks” the installation files, and configures the Windows Installer.
Follow the step by step instruction in the Windows Installer.
When prompted to restart your computer, click Yes. The InstallShield Wizard shuts down your computer, and restarts it.
The default port is 389 but you can also try using port 3268.
Port 3268 is an alternative testing port.
This feature is only intended to be used if you are migrating users from an older (4.x and prior) ShoreTel Converged Conferencing Server server to a newer (4.2 and above) Converged Conferencing server. This option is designed to protect those passwords during the migration.
No. The Conference Bridge is an appliance running on Linux, the only access will be loading the files, to make sure the filed been uploaded is not been infected, ShoreTel recommends to install AnitVirus on the machines have Administrator access to bridge to address the concern. There is no access in Conference Bridge to load AntiVirus.
For user who might have loaded virus-infected file for sharing, it will not harm Conference Bridge but it is users’ responsibility to have anti-virus program installed before downloading the sharing files..
The hostname for Converged Conferencing bridge needed to be a fully qualified domain name (or FQDN), which consists of a host and domain name, including top-level domain. For example, www.shoretel.com is a fully qualified domain name, www is the host, shoretel is the second-level domain, and .com is a top-level domain. The valid characters are letters, digits and hyphens.
You are installing Windows 2003 SP2 and would like to know if this is supported, and which ShoreTel builds.
The current version of ShoreTel Server supports Windows 2008 R2 64-bit. Please contact our support technicians for more detailed information regarding your installation.
As of ShoreTel 7.5 and below, Distributed Routing Service (DRS) allows larger systems to scale beyond 60 switches up to a total of 200 switches (including SoftSwitches). The Distributed Routing Service is optional on systems up to 60 switches, but must be enabled on systems with 60 or more switches. With ShoreTel 8 and above, ShoreTel supports up to 100 Switches per site with DRS disabled and 500 Switches per site with DRS enabled. Also, ShoreTel 8 also supports a Maximum of 100 V-Switches which count towards the total switch count per site.
When Distributed Routing Service is enabled, ShoreGear switches only exchange routing information with other switches configured in the same site, rather than exchanging information with every switch in the system. Although each ShoreGear switch only maintains routing information within its site, each ShoreWare server also includes an instance of the Distributed Routing Service, which maintains system-wide routing information. When site-to-site calls are initiated, ShoreGear switches contact the Distributed Routing Service in order to find the ShoreGear switch or switches necessary to complete the call.
In a system with more than one ShoreWare server, the ShoreGear switches may contact an alternate instance of the routing service if the primary instance is unreachable. ShoreWare servers have a hierarchical relationship, with the Headquarters server at the top of the hierarchy. As you add servers to the system using ShoreWare Director, you define the order of the servers in relation to the Headquarters server and the various sites in your system. Initially, the switches try to contact the nearest instance of the
Distributed Routing Service in the hierarchy. If that instance of DRS is unreachable, the switch contacts the instance of DRS at the parent server in the hierarchy as a fallback. If both instances of DRS are unreachable, the switch makes a best effort to route the call based on its internal routing tables built from communicating with peer ShoreGear switches at the same site. If the call is an external call, the call may be routed out a local trunk even though it may not be the lowest cost. If the call is an internal call, the call will be redirected to the Backup Auto-Attendant.
DRS OFF V- SWITCHES
DRS ON V- SWITCHES
> ST 10
Up to 100 Switches
Up to 500 Switches?
Up to 60 Switches
Up to 100 Switches
< ST 9.2 Up to 60 Switches Up to 200 Switches NA NA
You’re exporting records from the Shoreware CDR 6.1 database and you’d like to know what changes were made to the database in this version.
In the 6.1 release, the Extension field of Call Detail Record (CDR) is now a 16-character field. All subsequent columns have been shifted right by 9 characters.
Check that the local server time zone is correctly set in Director and in the Operating System. ShoreGear switches get their time offset from the site to which they are assigned.
Place the number in the Caller ID field of the Individual User. This number is used for caller ID on outbound calls. A caller ID entered here will take precedence over the user’s DID and the site’s CESID number. If no number is entered, the user’s DID or the site’s CESID will be used for outbound caller ID. NOTE: This feature is available only on outbound calls using a T1/E1 PRI trunk.
The server specifications for the Branch Office Solution is as follows:
Small Business Edition Integrated Server
800 MHz or better
512 MB RAM or better
40 GB hard disk or better
CD ROM or better
10/100 Ethernet NIC or better
One or more USB ports
No monitor, keyboard or mouse.
Microsoft Windows Server 2003 for Telecommunications Systems.
Small Business Edition (SBE) is designed for a single site implementation (one HQ site only), so the following features are not supported: AMIS, SMDI, On-net Dialing, and PSTN failover.
Benefits: Can be purchased with various bundles (Extension & Mailbox Licenses, etc…)
In Director, click Preferences under the Administration link in the navigation frame.
Play and Record – This will change how you play and record an auto-attendant prompt.
PC – Click this button if you will play and record via your PCs speakers and microphone. This requires that you have a sound card installed in your computer.
Telephone – Click this button if you will play and record via your telephone handset. In the Call Number field, you must enter a telephone number or the extension at which the system will call you.
The “Request System License Key” is disabled until the your first “keyed license” is entered. A system license key cannot be issued without corresponding user licenses to go with it (Extension+Mailbox, Extension Only, Mailbox Only, etc.). If you need assistance obtaining your licenses or a list of licenses purchased, please Contact TOTLCOM Support
Your system still has key(s) out of compilance so the system license key can’t be applied. Please contact your ShoreTel Partner or ShoreTel Installed Base Business Services Team – please Contact TOTLCOM Support to address any outstanding license issue.
KEYED LICENSES: these are licenses that we send specific encrypted keys for that are entered into the customers ShoreWare Director. Adding a license key causes the number of purchased licenses to increment by the number the key is generated for. Keyed licenses are:
All ShoreTel applications, including Workgroups, Voice Mail, and Account Code Collection, communicate via TAPI to other system components. All applications that need to interact with distributed call control do so via TAPI. Application use TAPI to communicate call control information to TMS, and TMS communicates this information to all other system components as needed. For example, whenever a user dials a number, the ShoreGear switch notifies TMS. TMS then presents the call infromation to the application via TAPI.
The Telco disabled Caller ID information; please check with Telco to make sure that they have not disabled the ability for you to send anything but the BTN “Bill To Number or Business Telephone Number”.
Media travels through the ShoreTel system using Real-Time Protocol (RTP). After call setup, media flows directly between IP phones via RTP. The ShoreGear switch is involved only when setting up or tearing down a call. A voice mail message is normal RTP traffic, unless it is a recorded voice mail message moving from one server to another.
Voice mail media streams conform to the G.711 codec. If a switch or IP phone is configured to use G.729 or ADPCM (for example, an intersite call), a media server proxy is used to transcode between G.729/ADPCM and G.711. Since the media server proxy is a switch resource, there are a limited number of G.729 proxies. If there are insufficient G.729 proxies, then ADPCM is used instead.
Endpoint location exchange is performed via ShoreTel’s proprietary Location Service Protocol (LSP). When switches first connect, they exchange all known SIP URLs. Afterwards, only configuration updates are transmitted. LSP is based on UDP. The service relies on keep-alive pings (sent every 30 seconds) to detect dead switches.
Upgrades can be performed in an on-line environment. Switches are updated when all ports are idle – new code is written to flash and rebooted on success.
For software upgrades, TOTLCOM simply installs the new software on the ShoreTel server, and all the ShoreGear voice switches, across all locations, are automatically and immediately to the new release.
In addition, users are notified of the new software release and will be prompted to automatically upgrade their software or administrators can easily upgrade the software on all client machines using Microsoft Active Directory Group Policies regardless of the permissions associated with those machines or the users who log into those machines.
ShoreTel Director software automates updates to the ShoreGear voice switches and ShoreTel Call Manager applications. Using an integrated software distribution mechanism, ShoreTel Director maintains an inventory of the firmware running on the ShoreGear voice switches, the ShoreTel IP Phones as well as the version of the ShoreTel Call Manager software, and automatically performs updates to keep the system running optimally.
All ShoreTel documentation is also refreshed on upgrades.
The ShoreTel IP solution is a single system distributed over multiple sites. Call control is distributed across the network via the ShoreGear voice switches that are deployed at each location. In addition, each ShoreGear switch contains the system routing configuration and performs its own call control enabling it to provide a single system that is distributed across all locations and is also individually survivable at each location.
ShoreTel software has passed a detailed evaluation and testing process managed by VMware and has achieved VMware Ready™ status.
With ShoreTel 12.3, there are several new platforms that have been tested for compatibility and now approved. This list includes the following:
TOTLCOM will custom design your system to leverage existing Virtual Machine Architecture and minimize additional hardware requirements.
All ShorePhone IP phones use Class 3 PoE and consume 4.3 W min idle and between 5.9 – 8.2 W max depending on model.
All IP Phones proposed have a built-in “headset” mode and jack. This allows answer and disconnect functions for headsets using a single dedicated button.
Standard headsets are available for all IP Phones proposed using a separate integrated headset jack and wireless headsets are available for the 655 and 565g models using Bluetooth 2.0.
Extension Assignment is a ShoreTel feature that allows users to reassign their extensions to any phone connected to the ShoreTel System or remote devices accessible to the network, including home phones and mobile devices.
Activating Extension Assignment from a system telephone assigns your extension to that device. When activating Extension Assignment from a remote device, you maintain an on-system extension presence by mapping your ShoreTel extension to the external device. ShoreTel Communicator displays calls that Extension Assignment forwards to your specified device. You can manage these calls in ShoreTel Communicator as if you received them through your home device. The device to which Extension Assignment assigns your extension rings when you receive a call. The active Call Handling Mode determines the disposition of unhandled calls. ShoreTel Communicator provides all call management functions except call answer. Extension Assignment calls must be answered on the assigned device. Voice calls made and received while Extension Assignment is active are placed in the extension’s call stack. Calls cannot be parked on an extension where Extension Assignment is active.
ShoreTel supports just about any analog phone sold in the market place. This can be corded or wireless. The analog telephones receive –48v dc power from the ShoreGear switch. For LCD/LED functions standard 110v ac power is required. In the event of power failure, the LCD/LED features will not be visible. No handset reconfiguration is required for analog phones. The administrator simply configures the user profile and settings in ShoreTel Director.
The ShoreTel system supports SIP trunk services natively however deployment with various Internet Telephony Providers may require a supplementary session border controller depending on the network environment.
The ShoreTel system natively supports SIP endpoints without requiring additional hardware or software. SIP Extensions give ShoreTel the ability to support many other vendors handsets. ShoreTel has worked jointly with other vendors to ensure interoperability such as Ascom, Multi-tech, Mediatrix and a few others. However many IPBX vendors typically won’t work together due to being competitors. ShoreTel has gone ahead and tested many of the other vendors handsets and typically they will work from a basic standpoint. However some features may not be supported.
TOTLCOM would like to share some important security information we have recently received from our partner ShoreTel. This information is directed towards all ShoreTel Customers and applies to all ShoreTel releases.
According to ShoreTel: A wave of attacks on phone systems in general and on voicemail systems in particular has been recently reported. These attacks have been reported via the media. Some of our customers have also reported attacks on voicemail.
Security vulnerabilities can create significant financial and privacy implications for a company due to unauthorized calls or modifications on the phone system. ShoreTel and TOTLCOM have the following recommendations and suggestions for improving the overall security of your ShoreTel deployments.
For systems on older ShoreTel builds you can search the Vmail-XXXXXX.log for “Log on failed password check” to check for any abnormal patterns.
Installing ShoreTel Communicator for Windows (before v13.2) on Windows 8 may fail; the client installer will roll back. There is no reliable work-around as of now.
We recommend using Communicator for Web as an alternative to Communicator for Windows on Windows 8.
Creating a new user via Director from IE10 on your Windows 8 PC will fail and corrupt the ShoreTel database!
A fix has been identified and will be in future builds of ShoreTel 12.3 and greater.
Please check with Totlcom before installing Communicator on Windows 8, and before using Windows 8 to manage your ShoreTel system.
Yes. All external inbound calls to a mobile device extension‐assigned from a ShoreTel extension will be recorded normally. To record external outbound calls from the extension‐assigned mobile device, the calls have to be placed via ShoreTel Communicator.
Yes, Mobility for iPad works with both iOS 5 and iOS 6.
Mobility is now optimized for iPad:
ShoreTel Mobility is a complete mobile solution that incorporates the most useful UC features into one application with support for a wider variety of PBXs. ShoreTel Mobility for iPad provides VoIP calling from internal Wi-Fi, as well as external
Hotspots with an automatic and secure SSL VPN. Presence and Instant Messaging is integrated into the Client. ShoreTel Mobility supports access via ShoreTel PBX, Cisco CUCM, Avaya Aura, Avaya Nortel CS1000, and TDM PBXs (via a TDM-to-
SIP voice gateway).
Unlike ShoreTel’s open, flexible support for a broad range of systems, Cisco Jabber for iPad is only supported on the Cisco PBX, requiring a large UC infrastructure with multiple system servers. Jabber does not have any kind of built-in VPN, which means users are required to manually initiate VPN or must have access to a Cisco ASA infrastructure. This has a negative impact on usability and productivity. Jabber accounts and user access must be configured on multiple systems, which wastes
time and resources. Cisco Jabber lacks a solution for BlackBerry and relies on a third-party solution, RIM MVS, compared to ShoreTel’s integrated support for Blackberry.
Avaya Flare relies on the Avaya PBX. Avaya Flare has VoIP calling (when connected to Wi-Fi) but does not have any cellular capability. The Avaya Flare Experience (the software that runs on the Flare tablet) can run on the iPad, but even then there is no integration with cellular so users can only place calls when connected to a Wi-Fi network. Avaya Flare does allow document sharing, which is a benefit that ShoreTel does not have integrated (however does have available separately through the ShoreTel Conferencing application). Like ShoreTel, presence status is shown for voice calling, video, and IM. Avaya lacks VoIP calling on other smartphone platforms such as Android and BlackBerry, while ShoreTel provides this support.
Yes, ShoreTel Mobility will automatically perform handover to cellular voice, when available, if the call quality degrades or the connection is lost. This enables users to focus on their communications without worrying about manually selecting the network that is utilized.
Yes, you can initiate a conference call for up to three lines by either:
(a) selecting the Conference button;
or (b) pressing and dragging a contact onto the call.
aaYou can add personal or enterprise contacts to a call.
A SIP Trunk is primarily a concurrent call that is routed over the IP backbone of a carrier using VoIP technology. SIP Trunks are used in conjunction with an IP-PBX and are thought of as replacements for traditional PRI or analog circuits. The popularity of SIP Trunks is due primarily to the cost savings of SIP, along with the increased reliability as backed by the SLAs of SIP Trunk Providers.
SIP Trunks are cheaper than analog circuits while maintaining the same service quality that businesses expect from line quality. SIP Trunks cost approximately $20 to $30 per trunk, versus $40 per analog circuit. In addition, long distance termination charges associated with SIP Trunks are much cheaper than traditional analog or TDM rates.
SIP Trunks realize their primary benefit over PRIs from cost savings. SIP trunks typically cost $20 to $30 per trunk for unlimited inbound and local calling along with a long distance rate that can be under 2 cents per call. When coupled with line oversubscription (e.g. a 30 person company purchasing just as many SIP trunks as they anticipate having concurrent calls…typically 8 to 10) SIP Trunks are a very cost effective way for a business to save money. Lastly, a primary benefit of SIP Trunks over PRIs is that SIP Trunks can be purchased in increments of 1, whereas PRIs have to be purchased in increments of 23 channels.
A business should look into purchasing SIP Trunks when they decide that their needs are best met with a premise-based system. This system is often referred to as an IP-PBX. Coupled with SIP Trunks, an IP-PBX serves up similar features to hosted solutions. SIP Trunks typically save a business customer money over a hosted solution in that a SIP Trunk can serve the needs of three to four employees (depending on oversubscription) while a hosted seat is needed for each employee.
SIP Trunks can work with a SIP-ready PBX. SIP Trunks can also be made to work with traditional analog or key systems with an IAD (Integrated Access Device). The SIP Trunk service provider will need to interoperate with the underlying equipment manufacturer. However, it should be noted that with the advent of standards around RFC 3261 and SIP Connect, the challenge of finding SIP Trunk Service Providers with SIP Trunk compatible equipment is significantly decreased.
SIP Trunks are virtual circuits delivered over an Internet Access line. Depending on the number of SIP Trunks purchased, and the amount of excess Internet connectivity, a business should consider purchasing more Internet Access. However, it’s important to know that when a SIP Trunk is not being used, the bandwidth otherwise allocated to a SIP Trunk is freed up for use in less intensive applications, such as e-mail and general web use. This dynamic allocation of bandwidth is yet another feature of SIP Trunks versus more traditional technologies, such as analog or PRI circuits.
NAT means Network Address Translation. When you connect to the Internet, information you send and receive is typically passed through a firewall, which protects your network infrastructure from hackers. Each individual network that is connected to the public Internet is identified by an IP address. Devices on the outside of your firewall see your publicly-routed IP Address; devices on the network inside of the firewall are addressed with private, non-routable IP addresses. When information passes through the firewall, it does so in the form of data “packets.” Each packet contains an IP address, telling it where it’s from and where it’s going. When a packet passes through the firewall to the public Internet, the private IP address is replaced with the publicly-routed IP address. When information passes from the public Internet through the firewall, the publicly-routed IP address is replaced with a private one and routed to the appropriate device.
SIP messages, which are carried inside of data packets, also include IP address information. Traditional NAT does not open the packet to change the IP information inside the SIP message. Therefore, the VoIP provider will not be able to send the call to the appropriate location. SIP requires a device to open the packet and adjust the IP information as it passes through the firewall.
Some firewall devices, known as Application Layer Gateways (ALGs), have the ability to make this change inside the SIP message. Though several ALGs are capable of SIP NAT traversal, TOTLCOM supports the Ingate SIParator router. Other firewalls, including Sonicwall and Cisco ASA/PIX, have this capability, but TOTLCOM can only provide technical support for the Ingate ALGs.
NAT-traversal problems manifest themselves in several easy-to-identify ways. If you are experiencing any of the following problems, you may have a NAT-traversal issue:
1. Are you having one-way audio?
2. Is the call setting up (ringing), but missing sound?
3. Are calls not setting up in the first place?
If you are experiencing any of these issues with your VoIP service, call TOTLCOM’s Customer Care Specialists at 1-800-300-5500.
TOTLCOM has several different options to help you deal with technical difficulties. If you are a Managed Network Services subscriber, we will take care of the problem from start to finish. If you don’t subscribe to MNS, call our Customer Care Specialists at 1-800-409-4357, options 1,1,1. If your network uses the Ingate SIParator or the Edgewater EdgeMarc ALGs, we should be able to troubleshoot your problems remotely. If you don’t currently use one of these TOTLCOM-supported ALG devices, there is another option. If you assign a public IP address directly to your phone system, you will obviate any NAT-traversal issues, because there is no translation necessary. This option does, however, leave your system vulnerable to attack. TOTLCOM does not recommend this option.
TOTLCOM supports two ALG devices: the Ingate SIParator and the Edgewater EdgeMarc router. Other devices capable of SIP packet translation are available, but TOTLCOM does not offer technical support for their use.
ALG stands for Application Layer Gateway. An ALG is a device that handles translation of the packet IP addresses (and for TOTLCOM VoIP service, the SIP IP information inside the packet) as they traverse the firewall. TOTLCOM supports the use of the Ingate SIParator ALG.
It is important to remember that ALG devices are designed to handle a specific number of concurrent calls. Be sure to purchase an ALG that fits your business needs. TOTLCOM’s Account Executives can help you determine which ALG is right for you. Call 800-300-5500 to speak with an Account Executive today.
Another name for ALG, a SIP-aware firewall is a device capable of handling the translation of SIP-packet IP addresses as the packets traverse the firewall. TOTLCOM provides technical support for the Ingate SIParator firewall for SIP-packet translation.
Far-End NAT traversal is a method of accomplishing NAT. See “What is NAT traversal?” for more information on NAT traversal. TOTLCOM does not utilize Far-End NAT Traversal, because doing so would introduce unnecessary delays and latency into our customers’ voice traffic. TOTLCOM places paramount importance on the voice quality of our service, and we feel that accomplishing NAT traversal on the customer’s premise is a far more effective and efficient technique.
QoS stands for Quality of Service. Quality of Service (QoS) is the idea that transmission rates, error rates, and other characteristics can be measured, improved, and, to some extent, guaranteed in advance. Is the sound clear? Is there any jitter or latency? These are questions that determine the level of QoS. At TOTLCOM, we pride ourselves on being an industry leader in QoS. Our engineers work tirelessly to ensure for our customers the highest possible level of QoS.
There are a several methods of ensuring the highest possible call quality. TOTLCOM recommends using a dedicated data connection of at least T1 speed, along with a properly-configured router using the latest QoS technologies.
TOTLCOM is dedicated to providing the highest possible level of call quality. To that end, our experienced engineers develop and test the latest QoS techniques. TOTLCOM VoIP service currently uses several techniques to ensure QoS, including TOS-splitting, traffic shaping, and voice-optimized internet (on some phone systems).
A dedicated voice connection is an Internet connection used solely for voice traffic. It carries no data packets, so traffic congestion, latency, and jitter are not issues. It is one of the best ways to ensure call quality, but it is generally not cost effective, because it requires the purchase of an additional T1-class Internet connection.
TOS stands for Type-of-Service splitting. It is a way of classifying and prioritizing the traffic on a data connection to give priority to voice traffic. If voice packets are held up by data traffic, latency and jitter issues can arise. Data packets, on the other hand, can handle slight delay far more easily. TOTLCOM currently utilizes TOS-splitting on outbound VoIP calls over Sprint’s and Qwest’s T-1 lines (Optional).
Traffic shaping is a technology used to enhance call quality. Traffic shaping software can differentiate between voice and data traffic. The software then “throttles back” the speed on the data traffic to make room for the voice traffic to proceed on the network unimpeded.
Voice Optimized Internet (VOA) is a technology TOTLCOM offers to ensure call quality. Essentially,one circuit (ADSL, SDSL, or T-1) is delivered to the customer site. The circuit is provisioned with two separate IP addresses one for data traffic and the other for voice traffic. Using this method, the voice traffic is prioritized from the customer premises all the way to the carrier network. This reduces the congestion for both inbound and outbound voice traffic.
VoIP traffic is as secure as any other data traffic you send out onto the public internet. There are ways of making the system more secure, including some technologies currently undergoing testing by TOTLCOM engineers, but none of the available technologies is yet universal. The best way to ensure security for your VoIP traffic is to employ the use of a SIP-enabled firewall.
Your VoIP system is as secure as any other data leaving or entering your network. If you employ the use of a SIP-enabled firewall, your VoIP traffic is relatively secure. For someone to gain access to your VoIP calls, they would have to steal all of the packets leaving your network. The only way to do that is to be physically present at one of the points of transmission.
TLS is software with the ability to handle secure signaling, including SIP signaling, using secure certificates similar to the way some web sites use certificates to ensure secure financial transactions. TOTLCOM engineers are currently testing TLS technology, but because it is not yet universal, it is not ready for implementation.
SRTP is a way to encrypt the signals carrying the voice information over the Internet. It is still in the development stage; TOTLCOM engineers are testing and monitoring its progress to determine whether it can be of use.
When a new call comes in, the line will flash, if you are not on a call, just pick-up the handset. The phone will automatically grab the ringing line for you. If you are on another call, just press the flashing line. The phone will automatically place your first caller on hold.
Pick up the handset, press a phone line that is not lit, and dial the number you wish to call. In addition you can dial 9 and the number you wish to call and the phone system will automatically grab the first available phone line to place your call on.
If you need to transfer a caller to another person, just press the CNF/TRN key. This places the caller on hold automatically. You will hear dial tone, dial the extension number, announce the call and hang-up. The call will ring on the Intercom button of the phone you transferred to.
To transfer a caller to someone’s voicemail, press the VM Transfer button, press their extension using the number buttons and press the #. Release Key If you are talking on a headset or using the speakerphone, the Release key will hang-up the line you are on.
To add another party to our conversation (conference), press the CNF/TRN button, the caller you were talking to will be placed on hold and you will hear dial tone. Dial either the extension number of the next party or dial the phone number of the person to add. After they pickup, you can press the CNF/TRN button again to include them into the call. If the party you wished to add is not available, just press the blinking line button to get back to your caller. You may have up to 8 people (including yourself) in a conference call, with a maximum of 6 outside callers.
To program a speed dial on your phone, enter programming mode (press #9876 on your phone). The phone will display “USER PROGRAMMING MODE.” Press the Spdial button, enter a speed dial number (100-119), 9 plus the phone number you wish to dialed (1-213-555-1212 for example), then press the Spdial button one more time to lock it in. To use a programmed speed dial, just press the Spdial button plus the number.
You can use this key to pickup a ringing extension within your designated group. Press the button. The caller will be transferred to you.
To program a one-touch button on your phone, enter programming mode (press #9876 on your phone). The phone will display “USER PROGRAMMING MODE.” Press the OTB button and enter the phone number you wish to dial (9-1-213-555-1212 for example), then press the OTB button one more time to lock it in. To use a programmed one-touch button, just press the OTB button.
Press this button to view the last number that called you. To view the next most recent call, press the down volume. To see the status of the call press the page button. To dial the call displayed, press 9 and the Caller ID button again.
To retrieve a parked call, if the light next to the button on your phone is red, simply press the button to retrieve the call. To retrieve a call parked on another extension, press the Page Pick-up button and key in the extension number (exp. 201).
To program the Call Forward button on your phone, enter programming mode (press #9876 on your phone). The phone will display “USER PROGRAMMING MODE.” Press the Call Forward button, enter the extension to forward all of your calls to, then press the Call Forward button one more time to lock it in. Press this key to forward all calls to the extension you programmed. The red light will be on to remind you that your calls are being forwarded. If the extension you forwarded calls does not answer and the call is forwarded to voice mail, the caller will be in your voice mail box! To turn off call forwarding, simply press the Forward All button again. For external call forwarding, program the button with a 91NNNXXXXXXX# for long distance or 9XXXXXXX# for local calls.
Press the Call Record button to start recording your conversation to your voicemail box. You can press this button to stop recording and to save the conversation or it will automatically save when you release the call (hang-up). The recording will appear as a new voicemail message
Press the Pause button to temporarily pause the recording of a phone conversation.
You can use this key to pickup either a ringing extension or a call that was placed on hold on another extension. Press the button and the extension number. The caller will be transferred to you.
Allows you to park a call on an extension or to retrieve a call parked. To park a call, with the caller on the line, press the park button and key in the extension. You will hear a confirmation tone. Hang up the call. To retrieve a parked call, if the light next to the button on your phone is red, simply press the button to retrieve the call. To retrieve a call parked on another extension, press the Park button and key in the extension number (exp. 201).
Press this key and speak into the handset. This will announce over all phones. Please use the Release Button to hang up.