Why Migrate?
From: Existing Avaya MERLIN® MAGIX Systems for firms with 20 – 200 employees
To: Avaya IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office
500 Professional Edition Communications Solutions
The Business Value of Migrating Now Avaya IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems fill the gap
between yesterday’s circuit switched PBXs and Key Systems and the IP-only solutions of tomorrow by meeting the total office communications needs of small-to-medium sized businesses.
• Take advantage of the most current technology – Most systems on the market today support Voice over IP capabilities that allow businesses to save money as well as deploy powerful new capabilities for end-users. MERLIN MAGIX systems were one of the best key/hybrid/PBX systems on the market in its time. As new systems have been introduced by Avaya,
MERLIN MAGIX has been discontinued as a current solution and eventually, parts may be difficult to obtain. The good news is that most of the phones supported on MERLIN MAGIX systems will migrate to IP Office systems, allowing your business to
retain their set investment.
• Automate message taking – Does your business have voice messaging capability today? It comes free with IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems and allows you to see all of your voice mail and e-mail messages in the same mailbox. A simple software upgrade allows your business to use the
messaging system to collect information from you callers, let your callers know how long their wait will be before they are helped, and record calls for enhanced customer service.
• Know how many new and old messages you have at a glance – With many of the large display sets available with IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems, Visual Voice shows users how many new and old voice mail messages they have, and provides the user with one-touch access to the voice
messaging system and several voice mail features. Visual Voice simplifies the ability to manage your voice messages. (Large displays sets only; Embedded Mail and VoiceMail Pro only)
• Make retrieving e-mail easy and lightweight – Stuck without a PC or laptop but don’t want loads of e-mail to come back to? Use the optional e-mail reader feature of VoiceMail Pro to hear e-mail messages read aloud. (MS Exchange only)
• Keep track of calls; know when you missed a call – Imagine being able to log calls using your PC screen to view incoming calls. Arm yourself with all of the information about your caller before you answer the phone. You can also see who is busy and who is free to take calls. Sound good? It is a free application provided to each user in the IP Office Small Office
Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems. A simple software upgrade allows for database integration and screen-pops. IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition system display sets will also log incoming, outgoing and missed calls and calls can be returned with the
press of a button.
• Enhance your collaboration and communications abilities – MERLIN MAGIX system users can establish a conference call with up to 5-parties. Using the built-in conferencing application that comes free with IP Office 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems, much larger conferences can be supported; up to 64-parties. In addition, you get a free Meet-me conference bridge. Set up a call, invite your attendees, and provide them a PIN number for security and save lots of time and money over using costly network conferencing services (VoiceMail Pro required).
• Save tons of money on external conferencing services with presentation viewing – Does your business frequently need to use services like Webex to have meetings and view presentations? The optional Conference Center application can be used to provide Web-based conferences and document viewing. Each user gets a unique PIN number to prevent unauthorized attendees (VoiceMail Pro required). Conference reservations can be made with e-mail invitations. The
conference host has audio control over which participants have Speak and Listen capabilities and those that have Listen-only privileges. (Not supported on IP Office Small Office Edition). Easy to use, easy to cost-justify!
• Reduce lost calls = reduce lost revenue - For customers waiting to reach an agent in a call group, demonstrate how VoiceMail Pro can tell the caller what number in line they are, and how long their wait time will be.
• Keep important recordings of calls to enhance customer service – Show the benefits of configurable call recording. Calls can be recorded by time of day, day of week, station number, agent ID number and more. All calls or just a percentage of calls can be automatically recorded. Calls recording can also be initiated manually. Advanced recording storage and retrieval applications are also available.
• Automatically get information from callers – Help the customer understand how they can automate informationgathering, collect revenue and run sales promotions using VoiceMail Pro’s Campaign Manager. Callers can provide typical information by speaking answers or by providing touchtone input to standard questions Campaign Manager is programmed
• Always be in touch – Discuss the value of VoiceMail Pro’s Personal Numbering “find me” service. Customers can enter details of where they can be located, including cell phone numbers, home numbers or alternate office locations. Calls to their number will follow them to their location, and if calls are busy or unanswered, the call will cover back to their VoiceMail Pro
mailbox so that all messages are stored in one location. The Voice Mail message can then be forwarded to an associated email account or alerts a predefined number to ensure messages are never missed. (T1/PRI required). Also, with optional Mobile Twinning, calls can ring simultaneously at your desk and at a home phone or cell phone with IP Office Small Office
Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems and if unanswered, cover back to the system voice mail. In addition, other users in the system get Busy Lamp Field indication on Mobile Twinned calls regardless of where the call is answered, improving communications even further.
• Branch Offices and Teleworking – The ability to work remotely, from home, a hotel or another office is becoming important to employees and employers in today’s 24x7 working world. Cost has always been a factor for employers. Now with IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems and optional SCN software,
remote working can become an inexpensive reality. Callers will never know if the employee is in the office or at their home. In addition, Branch Offices can now be easily tied into the main location using your data network, allowing your business to save money on long distance calls and offer a unified front to your customers.
• Networking – Does your business have multiple locations that are all independent today? IP Office Small Office Edition, 406 and 412 Release 4.0 and IP Office 500 Professional Edition systems allow you to optionally tie these locations together with full feature transparency between locations. With optional software, call center agents can login at remote offices and
employees can login to any phone in the network and the phone will take on that employee’s personal phone settings. These capabilities can represent a significant savings in toll calls using your existing data network, not to mention convenience and location freedom. Let us analyze your bills to see how much you can save!
Tuesday, August 21, 2007
CallTele wins Bartlett Fire
CallTele Chosen As Telecom Vendor For The Bartlett Fire Protection District
Elgin, IL – August 16, 2007 – CallTele, Inc. located in Elgin, Illinois, a network integrations company, specializing in business phone and data systems, voice and data cabling, high speed internet and phone services, has won the bid as the telecommunications equipment and service provider for the Bartlett Fire Protection District.
Aaron Kane, President of CallTele states, “we work closely with our customers to ensure that our business products and services are in alignment with their business and operational needs”. Captain Bill Gabrenya, who coordinated the telephone system purchase from CallTele, Inc. stated, “the new phone system will save the district $700 a month in phone costs, provide free calling between stations, provide email notification from voicemail messages and will increase the number of phone lines to 23 in preparation for bringing station three online next year.”
Elgin, IL – August 16, 2007 – CallTele, Inc. located in Elgin, Illinois, a network integrations company, specializing in business phone and data systems, voice and data cabling, high speed internet and phone services, has won the bid as the telecommunications equipment and service provider for the Bartlett Fire Protection District.
Aaron Kane, President of CallTele states, “we work closely with our customers to ensure that our business products and services are in alignment with their business and operational needs”. Captain Bill Gabrenya, who coordinated the telephone system purchase from CallTele, Inc. stated, “the new phone system will save the district $700 a month in phone costs, provide free calling between stations, provide email notification from voicemail messages and will increase the number of phone lines to 23 in preparation for bringing station three online next year.”
Monday, April 2, 2007
Move over AVAYA & CISCO…..Here comes Microsoft with Nortel riding shotgun!
Nortel Networks and Microsoft will further extend their Innovative Communications Alliance partnership into the carrier world on March 28 when the two introduce an integration of Nortel's carrier voice over IP softswitch with Microsoft's hosted applications platform.
The two vendors combined Nortel's Communication Server 2000, or CS 2000, softswitch with the hosted versions of Microsoft Exchange, Microsoft Office Live Communications Server and Windows SharePoint Services.
The combination will allow carriers to offer subscription services that bring together IP telephony, e-mail, collaboration, presence, instant messaging and desktop services.
At the same time, Nortel will provide integration and deployment services for carrier customers looking to implement a unified communications services offering based on the combined products.
Such hosted services are aimed primarily at SMBs (small and midsize businesses), a market which carriers have been looking to address, according to Sita Lowman, IMS market development leader at Nortel, in Richardson, Texas.
"Nortel is the prime systems integrator for carriers for this. We will provide full integration for them," she said.
For SMBs, the combined Nortel CS 2000 with Microsoft's Solution for Hosted Messaging and Collaboration bundle will deliver such IP telephony functions as click- to-call, traditional voice communications, and inbound and outbound dialing from the Microsoft Office Communicator desktop client. Users will also be able to place and receive calls, and access their calendars and contacts from their PC or desktop telephone from their office, home or hotel room.
Aaron Kane
www.callteleusa.com
The two vendors combined Nortel's Communication Server 2000, or CS 2000, softswitch with the hosted versions of Microsoft Exchange, Microsoft Office Live Communications Server and Windows SharePoint Services.
The combination will allow carriers to offer subscription services that bring together IP telephony, e-mail, collaboration, presence, instant messaging and desktop services.
At the same time, Nortel will provide integration and deployment services for carrier customers looking to implement a unified communications services offering based on the combined products.
Such hosted services are aimed primarily at SMBs (small and midsize businesses), a market which carriers have been looking to address, according to Sita Lowman, IMS market development leader at Nortel, in Richardson, Texas.
"Nortel is the prime systems integrator for carriers for this. We will provide full integration for them," she said.
For SMBs, the combined Nortel CS 2000 with Microsoft's Solution for Hosted Messaging and Collaboration bundle will deliver such IP telephony functions as click- to-call, traditional voice communications, and inbound and outbound dialing from the Microsoft Office Communicator desktop client. Users will also be able to place and receive calls, and access their calendars and contacts from their PC or desktop telephone from their office, home or hotel room.
Aaron Kane
www.callteleusa.com
Tuesday, February 20, 2007
VoIP Secure or not Secure.. That is the question
As VOIP systems proliferate, so, too, must the measures taken to secure them. Luckily for IT administrators, several resources are available to help them do just that.
In the book "Hacking Exposed VOIP: Voice over IP Security Secrets & Solutions," for example, authors David Endler (director of security research at TippingPoint) and Mark Collier (chief technology officer of SecureLogix) bring to life the imminent threat of VOIP attacks, describing in detail how an attacker could discover, enumerate, probe and eventually co-opt an existing voice network.
Moreover, the book provides a useful starting place for VOIP adopters to begin shoring up their own networks.
The $50, 539-page book is a must-read for IT administrators, particularly those who are managing a voice network but are not totally comfortable with the technology—and are perhaps relying too much on resellers for the stability and security of the network.
Endler and Collier also have created several tools to automate voice-specific scans and penetration attacks of commonly used end-user devices and VOIP infrastructure components. These tools, along with Google hacking tips and a database of stock voicemail recordings, can be found on the book's companion Web site.
Another good resource is VOIPSA, or Voice over IP Security Alliance, which has put together a number of useful tools. The Threat Taxonomy, for example, enumerates known types of attacks and organizes them into general categories.
These include social threats, eavesdropping, or interception and modification. The group also maintains a best-practices mailing list that is still in the organizational stage but holds great promise. Data and voice ties.
The health of a VOIP network is tied to the health of the data network. Beyond throughput and latency demands, VOIP relies heavily on several legacy data services to operate correctly. For example, VOIP administrators should think about shoring up services such as DNS (Domain Name Service), DHCP (Dynamic Host Configuration Protocol) and TFTP (Trivial File Transfer Protocol)—bedrocks of the network that may have been forgotten because, in most circumstances, they just work.
All of that said, isolating voice traffic should be a priority, as malware-infested desktops or servers could be used as a launchpad for attacks against a voice system that is not otherwise exposed to Internet traffic. Maintaining separate VLANs (virtual LANs) for voice and data traffic is a worthy endeavor: It will isolate VOIP components to some extent from other endpoints, with the added benefit that QOS (quality of service) will be easier to implement.
However, VOIP implementers may have to make some hard decisions about the role of IP softphones in the network because the workstations on which they are installed should be separated from the voice network, too.
Assuring call privacy through the use of encryption (of the voice payload, call signaling or both) has been a nonstarter for many VOIP adopters because call quality has traditionally trumped security as the primary objective. Many common voice-quality monitoring and assessment tools aren't effective on an encrypted stream, and IT staffers certainly can't record and play back encrypted voice calls to manually ensure call quality.
However, some new assessment tools are coming down the pike that can estimate call quality by measuring factors that can be derived from an encrypted stream, such as latency and jitter. For example, at the RSA Conference in San Francisco Feb. 5-9, AirMagnet demonstrated its AirMagnet VoFi Analyzer, which includes the ability to derive an MOS (mean opinion score) and R-values from metrics gathered without actual access to call payload.
Any of the tools mentioned here will help administrators lock down their VOIP systems, but, in general, admins must become more acquainted with the tricks of the trade. Just as they learned to use penetration testing and assessment tools for the data network, administrators must grow their skills to adapt to the vicissitudes of VOIP.
In the book "Hacking Exposed VOIP: Voice over IP Security Secrets & Solutions," for example, authors David Endler (director of security research at TippingPoint) and Mark Collier (chief technology officer of SecureLogix) bring to life the imminent threat of VOIP attacks, describing in detail how an attacker could discover, enumerate, probe and eventually co-opt an existing voice network.
Moreover, the book provides a useful starting place for VOIP adopters to begin shoring up their own networks.
The $50, 539-page book is a must-read for IT administrators, particularly those who are managing a voice network but are not totally comfortable with the technology—and are perhaps relying too much on resellers for the stability and security of the network.
Endler and Collier also have created several tools to automate voice-specific scans and penetration attacks of commonly used end-user devices and VOIP infrastructure components. These tools, along with Google hacking tips and a database of stock voicemail recordings, can be found on the book's companion Web site.
Another good resource is VOIPSA, or Voice over IP Security Alliance, which has put together a number of useful tools. The Threat Taxonomy, for example, enumerates known types of attacks and organizes them into general categories.
These include social threats, eavesdropping, or interception and modification. The group also maintains a best-practices mailing list that is still in the organizational stage but holds great promise. Data and voice ties.
The health of a VOIP network is tied to the health of the data network. Beyond throughput and latency demands, VOIP relies heavily on several legacy data services to operate correctly. For example, VOIP administrators should think about shoring up services such as DNS (Domain Name Service), DHCP (Dynamic Host Configuration Protocol) and TFTP (Trivial File Transfer Protocol)—bedrocks of the network that may have been forgotten because, in most circumstances, they just work.
All of that said, isolating voice traffic should be a priority, as malware-infested desktops or servers could be used as a launchpad for attacks against a voice system that is not otherwise exposed to Internet traffic. Maintaining separate VLANs (virtual LANs) for voice and data traffic is a worthy endeavor: It will isolate VOIP components to some extent from other endpoints, with the added benefit that QOS (quality of service) will be easier to implement.
However, VOIP implementers may have to make some hard decisions about the role of IP softphones in the network because the workstations on which they are installed should be separated from the voice network, too.
Assuring call privacy through the use of encryption (of the voice payload, call signaling or both) has been a nonstarter for many VOIP adopters because call quality has traditionally trumped security as the primary objective. Many common voice-quality monitoring and assessment tools aren't effective on an encrypted stream, and IT staffers certainly can't record and play back encrypted voice calls to manually ensure call quality.
However, some new assessment tools are coming down the pike that can estimate call quality by measuring factors that can be derived from an encrypted stream, such as latency and jitter. For example, at the RSA Conference in San Francisco Feb. 5-9, AirMagnet demonstrated its AirMagnet VoFi Analyzer, which includes the ability to derive an MOS (mean opinion score) and R-values from metrics gathered without actual access to call payload.
Any of the tools mentioned here will help administrators lock down their VOIP systems, but, in general, admins must become more acquainted with the tricks of the trade. Just as they learned to use penetration testing and assessment tools for the data network, administrators must grow their skills to adapt to the vicissitudes of VOIP.
About CallTele, Inc
CallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP. What solutions are provided by CallTele? Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
847-888-1900
Thursday, February 1, 2007
Nortel and Microsoft equals SMB VoIP
Microsoft and Nortel Networks on Jan. 17 revealed more about their Unified Communications Alliance, announcing real deliverables and detailing a road map for further products due next year.
The alliance, launched in the summer of 2006, has yielded over the last six months a series of integrated offerings due out this year. First of the new deliverables is to be a joint Unified Messaging offering that provides interoperability between Nortel's Communication Server 1000 and Microsoft Exchange Server 2007. It represents the first integration of Exchange with Nortel's native implementation of the industry standard SIP (Session Initiation Protocol). Nortel will supplement the offering with its professional services for design, deployment and support.
The alliance, launched in the summer of 2006, has yielded over the last six months a series of integrated offerings due out this year. First of the new deliverables is to be a joint Unified Messaging offering that provides interoperability between Nortel's Communication Server 1000 and Microsoft Exchange Server 2007. It represents the first integration of Exchange with Nortel's native implementation of the industry standard SIP (Session Initiation Protocol). Nortel will supplement the offering with its professional services for design, deployment and support.
More info Click here: http://www.eweek.com/article2/0,1895,2084997,00.asp
About CallTele, IncCallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP. What solutions are provided by CallTele? Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery Aaron KaneVOIP ExpertCallTele, Inchttp://www.callteleusa.com/847-888-1900
Friday, January 19, 2007
Benefits of Cisco Unified Communications Solution
An end-to-end, standards-based Cisco Unified Communications Solution brings together data, voice, and video communications on a single network to deliver more effective communications that can help lower operating costs, increase customer satisfaction, and optimize business productivity.The solution enables companies to quickly reduce their total cost of network ownership by consolidating their voice and data systems to a converged IP network and allowing customers to integrate with critical business applications such as Microsoft CRM and Microsoft Outlook. Using a single network for all communications helps them reduce monthly toll and leased-line charges, consolidate separate maintenance contracts for voice and data systems, and simplify cabling and wiring requirements.The Cisco Unified Communications Solution can also deliver significant strategic benefits by helping a business improve its operational efficiency with communications services like IP call handling and unified messaging. A recent study by The Radicati Group, Inc., a research firm in Palo Alto, California, estimates that unified messaging generates 25 to 40 minutes per day of additional productivity per employee. Since the Cisco Unified Communications Solution provides support for wireless and mobile workforces, staff can stay connected and productive, regardless of location. Employees can spend more time with customers or managing business.Combining separate data and voice networks into one IP infrastructure enables companies to simplify operations as they build additional value into their networks. With the Cisco Unified Communications Solution, companies can merge previously disparate support and administrative staffs for voice and data networks. By eliminating redundancy, companies can redirect their assets toward strategic core business investments. Many organizations are significantly improving their network support staff productivity as a result of deploying a converged IP network. With the Cisco Unified Communications Solution, companies can also support innovative XML and business applications, helping to ensure content integration with back-end database systems for sales and marketing, new service creation, and finance, legal, and technology operations. For more information click here
About CallTele, IncCallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
What solutions are provided by CallTele?
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
What solutions are provided by CallTele?
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
847-888-1900
Sunday, January 14, 2007
Nortel BCM 50
Business Communications Manager 50 is an all-in-one, affordable platform for converged voice and data communications for small to medium business with 3 to 20 users, yet scalable to serve more than 40. Business Communications Manager 50 provides advanced capabilities such as robust telephony features, voice messaging and unified messaging, IP networking, Internet access, contact centers with skills-based routing, and IP telephony. With an easy-to-use and highly flexible architecture, the Business Communications Manager 50 enables small sites to benefit from convergence capabilities that were previously only available to much larger organizations.
About CallTele, Inc
CallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
CallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
What solutions are provided by CallTele?
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
847-888-1900
Monday, January 8, 2007
What is SIP and how will it work with VOIP
A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signaling. However, it has been designed to enable the building of such features in network elements known as Proxy Servers and User Agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.
SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signalling System 7 (SS7), though the two protocols themselves are very different. SS7 is a highly centralized protocol, characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol. As such it requires only a very simple (and thus highly scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) as opposed to traditional SS7 features, which are implemented in the network.
Although many other VoIP signaling protocols exist, SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. SIP has been standardized and governed primarily by the IETF while the H.323 VoIP protocol has been traditionally more associated with the ITU. However, the two organizations have endorsed both protocols in some fashion.
SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP acts as a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what IP ports to use, the codec being used etc. In typical use, SIP "sessions" are simply packet streams of the Real-time Transport Protocol (RTP). RTP is the carrier for the actual voice or video content itself.
The first proposed standard version (SIP 2.0) was defined in RFC 2543. The protocol was further clarified in RFC 3261, although many implementations are still using interim draft versions. Note that the version number remains 2.0.
SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. SIP shares many HTTP status codes, such as the familiar '404 not found'. SIP proponents also claim it to be simpler than H.323. However, some would counter that while SIP originally had a goal of simplicity, in its current state it has become as complex as H.323. Others would argue that SIP is a stateless protocol, hence making it possible to easily implement failover and other features that are difficult in stateful protocols such as H.323. SIP and H.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future, unrealized applications.
Hardware endpoints — devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication — are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) or DUNDi to translate existing phone numbers to SIP addresses using DNS, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it.)
Today, software SIP endpoints are common. Microsoft Windows Messenger uses SIP. iChat AV, Apple Computer's AOL Instant Messenger compatible client, has supported audio and video chat through SIP, first in the 2003 public beta and now in the production version. On Linux, Ekiga supports SIP.
SIP also requires proxy and registrar network elements to work as a practical service. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peer-to-peer, this approach is impractical for a public service. There are various softswitch implementations (by Nortel, Sonus and many more) that can act as proxy and registrar. Other companies, led by Ubiquity Software and previously Dynamicsoft (who are now no more, after being acquired by Cisco in 2004) have implemented products based on the proposed standards, building on the Java JAIN specification. These follow the SIP Servlet API, JSR 116. These products allow deploying applications of arbitrary complexity onto a telephony network. In particular, they play the role of Application Servers in the IMS architecture.
SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signalling System 7 (SS7), though the two protocols themselves are very different. SS7 is a highly centralized protocol, characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol. As such it requires only a very simple (and thus highly scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) as opposed to traditional SS7 features, which are implemented in the network.
Although many other VoIP signaling protocols exist, SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. SIP has been standardized and governed primarily by the IETF while the H.323 VoIP protocol has been traditionally more associated with the ITU. However, the two organizations have endorsed both protocols in some fashion.
SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP acts as a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what IP ports to use, the codec being used etc. In typical use, SIP "sessions" are simply packet streams of the Real-time Transport Protocol (RTP). RTP is the carrier for the actual voice or video content itself.
The first proposed standard version (SIP 2.0) was defined in RFC 2543. The protocol was further clarified in RFC 3261, although many implementations are still using interim draft versions. Note that the version number remains 2.0.
SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. SIP shares many HTTP status codes, such as the familiar '404 not found'. SIP proponents also claim it to be simpler than H.323. However, some would counter that while SIP originally had a goal of simplicity, in its current state it has become as complex as H.323. Others would argue that SIP is a stateless protocol, hence making it possible to easily implement failover and other features that are difficult in stateful protocols such as H.323. SIP and H.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future, unrealized applications.
Hardware endpoints — devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication — are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) or DUNDi to translate existing phone numbers to SIP addresses using DNS, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it.)
Today, software SIP endpoints are common. Microsoft Windows Messenger uses SIP. iChat AV, Apple Computer's AOL Instant Messenger compatible client, has supported audio and video chat through SIP, first in the 2003 public beta and now in the production version. On Linux, Ekiga supports SIP.
SIP also requires proxy and registrar network elements to work as a practical service. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peer-to-peer, this approach is impractical for a public service. There are various softswitch implementations (by Nortel, Sonus and many more) that can act as proxy and registrar. Other companies, led by Ubiquity Software and previously Dynamicsoft (who are now no more, after being acquired by Cisco in 2004) have implemented products based on the proposed standards, building on the Java JAIN specification. These follow the SIP Servlet API, JSR 116. These products allow deploying applications of arbitrary complexity onto a telephony network. In particular, they play the role of Application Servers in the IMS architecture.
Who is CallTele
CallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
What solutions are provided by CallTelePhone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
847-888-1900
Thursday, January 4, 2007
Avaya, the Leader in Call Center Solutions
A new report from Gartner Inc. on the 2005 global contact center market1 shows that Avaya Inc. (NYSE: AV) is the market leader with 39.1% of all contact center agents – more than triple the share of the company's closest competitor. The Gartner report also shows that Avaya leads each region of the global contact center market, including:
Asia Pacific with a 54% share (44.4 points ahead of the closest competitor).
North America with a 41.1% share (28.1 points ahead of the closest competitor).
Latin America with a 32.8% share (10.1 points ahead of the closest competitor).
Europe, the Middle East and Africa with a 31.4% share (more than 20.7 points ahead of the closest competitor).
A leading global provider of business applications, systems and services, Avaya has one of the most comprehensive contact center portfolios on the market. Solutions range from hosted contact center services for companies with as few as 30 agents to sophisticated contact center applications that can be scaled to support thousands of agents handling millions of customer interactions daily.
One of the companies using Avaya contact center solutions to deliver personalized, market-differentiating services is Banco Santander Chile, the country's largest bank with more than 300 branches and 2 million customers.
Banco Santander's more than 300-agent contact center is powered by Avaya Communication Manager call processing software running on an Avaya S8700 Media Server. An Avaya Call Management System provides the information and management tools the bank needs to monitor and analyze the performance of its contact center operations, showing where improvements are needed and where it can take fast effective action. Banco Santander agents answer calls with CallMaster® Digital Voice Terminals specifically designed by Avaya to improve the productivity of high-volume contact centers.
With the Avaya solution in place, Banco Santander says it has improved the productivity of its agents and both the quality and efficiency of its interaction with customers.
Asia Pacific with a 54% share (44.4 points ahead of the closest competitor).
North America with a 41.1% share (28.1 points ahead of the closest competitor).
Latin America with a 32.8% share (10.1 points ahead of the closest competitor).
Europe, the Middle East and Africa with a 31.4% share (more than 20.7 points ahead of the closest competitor).
A leading global provider of business applications, systems and services, Avaya has one of the most comprehensive contact center portfolios on the market. Solutions range from hosted contact center services for companies with as few as 30 agents to sophisticated contact center applications that can be scaled to support thousands of agents handling millions of customer interactions daily.
One of the companies using Avaya contact center solutions to deliver personalized, market-differentiating services is Banco Santander Chile, the country's largest bank with more than 300 branches and 2 million customers.
Banco Santander's more than 300-agent contact center is powered by Avaya Communication Manager call processing software running on an Avaya S8700 Media Server. An Avaya Call Management System provides the information and management tools the bank needs to monitor and analyze the performance of its contact center operations, showing where improvements are needed and where it can take fast effective action. Banco Santander agents answer calls with CallMaster® Digital Voice Terminals specifically designed by Avaya to improve the productivity of high-volume contact centers.
With the Avaya solution in place, Banco Santander says it has improved the productivity of its agents and both the quality and efficiency of its interaction with customers.
Who is CallTeleCallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
What solutions are provided by CallTele
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
http://www.callteleusa.com/
847-888-1900
What solutions are provided by CallTele
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
http://www.callteleusa.com/
847-888-1900
Wednesday, January 3, 2007
Avaya One-X SIP VoIP Solution
Avaya one-X™ Quick Edition delivers intelligent communications to small offices. SIP-based peer-to-peer technology means set-up is simple, with powerful options to customize. Quick Edition delivers business class communication features including voice mail and auto-attendant and rich call management features including Call Forward, Conferencing, Park, Page and Retrieve to create a professional business image. Quick Edition can connect multiple remote branches and enables central management. Now even small offices can equip home workers with the same features they have in the office.
Product Details Quick Edition eliminates the need for a central server without eliminating performance. Embedding communication software in the phones removes cost and increases reliability. It is scalable, too. As you add employees, simply add additional telephones. Should your business needs change, in the future Quick Edition phones will be upgradeable to more advanced Avaya solutions. Quick Edition truly provides a simple, cost-effective, and connected business communication system.
Who is CallTeleCallTele is a business to business telecommunication company, specializing in business phone system sales, data equipment sales, local, long distance and high speedinternet phone services, voice and data architectural design, disaster recovery and VoIP.
What solutions are provided by CallTele
Phone Systems, Data Equipment and Communication Accessories• Maintain and Service Voice and Data Networks, Local and Long Distance Phone Service / High Speed Internet Service Nationwide Dispatch for Installation, Service, Moves, Adds or Changes, VoIP Solutions and Disaster Recovery
Aaron Kane
VOIP Expert
CallTele, Inc
847-888-1900
Monday, January 1, 2007
Nortel offers small to large phone systems.
Happy New Year from CallTele and its team of experts.
Nortel CiCS
Designed for growing businesses, the Norstar Compact Integrated Communications System (CICS) offers from 4 to 8 lines and 8 to 24 telephone extensions. It supports all Norstar and Business Series telephones, including fully-featured digital functions with LCD and soft keys. Easily add Norstar applications, such as voice mail, automated attendant, computer telephony integration (CTI), and automatic call distribution (ACD).
Nortel MICS
The Norstar Modular Integrated Communications System (ICS) is a key system designed for growing small-to-medium businesses and is expandable up to 248 ports. Advanced integrated applications include voice messaging, unified messaging, digital networking, enhanced call centers and speech recognition. The Modular ICS can grow with your business through the addition of trunk and station modules as you need them.
CallTele, a leader in the Technology Sector, is a communications company that delivers a variety of products for all your voice and data needs. With expertise ranging from Voice over IP (VoIP) to traditional telephony, CallTele will design, implement, and service your organization both locally and nationally.
CallTele's customer service center supports an array of products and applications across multiple product platform environments. Our skilled and trained customer service staff will project manage and service your requirements on a national level. CallTele's certified network of contractors are available 24 hours a day 7 days a week. CallTele's national foot print provides you with one contact for all your product and service needs.
A provider of:Avaya, Nortel, Siemens, Panasonic, Plantronics, GN Netcom, Polycom, Cisco, Extreme Networks
Aaron Kane
VoIP Expert
CallTele, Inc
847-888-1900
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