Sunday, July 17, 2011

Redfone foneBridge 2

Redfone makes hardware and software products for open-source VoIP implementations, including bridges and gateways. The foneBridge 2 itself is a T1/E1 PRI-to-Ethernet bridge enabled for Asterisk, trixbox, FreeSwitch, Elastix and other platforms, providing Ethernet communication using Asterisk TDMoE (Time-division multiplexing over Ethernet) on Asterisk servers, instead of T1/E1 connections. The appeal of TDMoE for customers is that it offers traditional TDM features over Ethernet instead of pricey TDM hardware.

Freeside Billing

Freeside's open-source billing, CRM, trouble-ticketing and automation software copped a Best of Show award at this month's Digium Asterisk World / IT Expo in Miami -- the second year in a row it's collected the honor. It's as described: open-source software for ISPs, VoIP, hosted solutions, service providers and other customers, including Web-based businesses, that don't want to pay through the nose for comparable proprietary systems. Freeside itself began as an internal application developed by a regional ISP in the mid-90s, from which its founders released the first open-source version of its code and turned the application into a consultancy and provider of pre-configured hardware.

OrecX Oreka TR

OrecX makes voice recording, screen capture and quality monitoring products to be used with call centers. Its basic Oreka GPL recorder integrates with any phone system and computer and is free. The total recorder (TR) is available for $199 a year and is OrecX's value-add version, offering the ability to record VoIP SIP sessions through network packets, record multiple channels from standard sound devices like microphones, and segment and split audio into different, retrievable files. According to OrecX's, call replay is accessed through the Web and available on all standard browsers without a software upgrade.

Friday, July 15, 2011

Fonality trixbox Pro

Fonality's trixbox Pro, the business version of its IP-PBX software trixbox, comes in four editions: Pro Standard Edition (SE), Enteprise Edition (EE), Call Center Edition (CCE) and Unified Agent Edition (UAE). Fonality uses what it calls a hybrid-hosted architecture: a split, hosted-plus-premise model that includes free VoIP calling and monitoring like most hosted solutions provide, but also PSTN connectivity and call privacy like you'd find in a more traditional PSTN set-up. Started by enterpreneur Chris Lyman about five years ago, Fonality first came to many solution providers' notice with its Asterisk-based PBXtra.

Asterisk Dialogic DiaStar

Dialogic's DiaStar Server is a platform for app dev -- Asterisk-based or otherwise -- around functions like call progress analysis, fax, fax over IP, SS7 and Sigtran. The goal, according to Dialogic, is to get developers as much flexibility as possible, whether they want to leverage PSTN protocols like ISDN, SS7 or CAS or provide rich applications around voice and video. For LAN environments, Dialogic uses Woomera, the open-source IP-based signaling protocol.

Wednesday, July 13, 2011

Asterisk Synway

Asterisk based IP-PBXs are becoming more available and popular on the PBX business market, threatening the traditional proprietary PBX manufacturers position continuously. IP-PBX has proved itself to be efficient because it can reside on the current network using existing LAN. Besides, it can greatly lower operational costs especially for systems with a higher number of users.

Synway specializes in providing superior Media Processing & Signaling Technologies, Telephony Hardware and Integrated Multimedia Switch in use for convergence (voice/data/video) communications. Since 1995, over 1000 software developers and system integrators have integrated Synway's offerings to deliver a broad range of TDM and VoIP-based applications, including unified communications, call center, mobile VAS, media gateway, fax, conferencing, call recording, Asterisk-based open source applications for operators and enterprises worldwide.

Asterisk Skype

Skype for Asterisk is an add-on channel driver for Asterisk-based PBX systems. Adding Skype for Asterisk to any Asterisk server enables complete access to the Skype community, including low cost PSTN access and calling to more than 500 million registered Skype users. Use of Skype for Asterisk requires a Skype Manager account.

Key Features

• Make Skype to Skype calls
• Calls to landlines and mobile phones
• Receive calls with SkypeIn
• Make world-wide PSTN calls with SkypeOut
• Make and receive multiple concurrent Skype calls from the same Skype account
• DTMF support for incoming and outgoing calls
• Read Skype profile fields from incoming calls
• Set and retrieve online status
• Set privacy settings
• Handle incoming Skype calls using all Asterisk applications (voicemail, ACD, MeetMe conferencing, etc.)
• Simultaneous access from both Asterisk and the Skype desktop client
• Use Asterisk phone for voice and Skype desktop client for IM, video
• Supports G.711 and G.729 (included) codecs
• Send and receive text chat messages on the Skype network

Asterisk PRI

PRI is the standard for connections to offices. It's based on a T1 line in the US,
and E1 line in Europe. The T1 PRI consists of 24 channels, the E1 PRI of 32.

PRI has varying number of channels depending on the country.
In North American and Japan it's 23xB + 1xD (23 64Kbps digital channels + 1 64Kbps signalling/control channel) on a T1 1.544 Mbps
In Europe and Australia it's 30xB + 2xD on an E1 2.048 Mbps
(One timeslot on the E1 is not available for the user since it is used for internal synchronization purposes.)

PRI is also available in partial variants - ie 10xB + D.

Question: What about outside US and Europe - what's being used?
Answer: Most of the rest of the world use E1. Japan uses a variant of T1.


NFAS (Non Facility Associated Signalling) allows sharing one D channel accross multiple PRI.

As well as 'standard' signalling there are vendor specific extentions to PRI. The relevant standards include:

National ISDN v1 aka. NI-1
National ISDN v2 aka. NI-2 (see SR-4994)
Nortel DMS100, DMS250 (see NIS-A211-1)
Lucent 4ESS, 5ESS (see TR 41459)
some recent versions of the 5ESS support protocol extensions that Lucent refers to as Natianal ISDN v3 aka. NI-3

Tuesday, July 12, 2011

Telemarketing Techniques

A ringing phone is the result of your marketing efforts, which you pay good money to develop and implement and the fact that your phone rings indicates that you’re marketing is working. Don’t squander your hard-earned marketing dollars (and reputation) by underestimating the value of each phone call.

The value in any business is its customer and prospect list and the fastest way to build that list is to ask for contact information.

Have you heard - - cold-calling doesn’t work - - but “warm-calling” does. Warm-calling happens when you call someone who has already had some type of experience with you. Whenever you call a prospect, have a valid reason to call.

People call you because they want something. Let me say that again, people call you because they want something. They might want a price, an explanation, to report a problem, to speak with a manager etc. Whatever the case may be, they want something.

The key to selling is to give people what they want. What better time to listen to what somebody wants than when they call you up? If you listen well enough, your prospects will tell you just what you need to know to sell them your product or service.

One of the most common challenges to telephone marketing is how to handle price checkers, people who call you up just to get a price. This is especially dangerous if they have not yet received enough information to determine value.

It leaves you to compete price alone. This leaves you at a disadvantage unless you sell a bottom-end, cheap-as-dirt product. If you do, hopefully you have a cost advantage over your competitors.

Perhaps the most detrimental result of giving out your price over the phone is that you have just taken away the single most powerful motivation for people to come into your business and speak with you personally. You’ve just set yourself up to be price shopped.

A phone call without an appointment is like building a website and not getting visitors. Why work to make your phone ring if you’re not going to try and get an appointment? You should ask for an appointment on most every call.

There’s no better time to market to somebody then when they are waiting to speak to you. Instead of telling the person waiting on the phone about how great you are, help them to solve their problems. Here is a typical message that I hear often: “Thanks for calling ABC Pools, our store hours are from 9:00 a.m. to 6:00 p.m. Please call back during our normal business hours. Thank you.” Click.

Your telephone can be a powerful marketing tool if used correctly. With some know-how training you’ll be able to see results immediately.

CoreDial's VoiceAxis

CoreDial's VoiceAxis is a turnkey software package targeted to enterprise-level businesses and service providers. The suite is designed for the management, billing and provisioning of hosted PBX, VoIP, SIP trunking and other related products -- essentially a one-stop management tool that Asterisk lovers have embraced for its ease-of-use with Asterisk PBXes. Version 3.0 of VoiceAxis arrived in October 2009.

Aheeva's Asterisk-based Contact Center Suite (CCS)

Aheeva's Asterisk-based Contact Center Suite (CCS) 2.2 is designed as an all-in-one contact center offering that ties together call routing, contact management functions and support and research functions -- all of the traditional call center features -- and latches on to an existing IP infrastructure to provide those things. According to Aheeva, it's fully scaleable and requires only a Web browser for access, which ramps up its appeal as a services play for remote monitoring, management and network analytics. Aheeva's message is that it's bringing functionality of a traditional telephony switch-based call center at a fraction of what proprietary systems cost.

Aksys' KONNECT Office Phone System

Aksys' KONNECT Office Phone System copped a best of show award at last year's IT Expo, the umbrella conference that includes Digium Asterisk World. It's presently available in two models: the KONNECT 600PL VoIP and landline phone and the 600P IP phone, which includes the 600PL's features minus a landline (FXO connector), and includes three-way conference calling, public address, paging to multiple phones, 3-digit dialing between sites and the ability to support as many as 14 concurrent calls per phone. Aksys markets KONNECT as an "office phone system inside a phone" and has gained traction with small businesses looking to integrate VoIP and landline functions and home and remote offices.