SIP Phones

maple4VOIP DE

July 3, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Star2Star Communications Reviews

June 30, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Founded in 2006, Star2Star has earned its status as a respectable VoIP provider in the market. Although its services extend to the entire United States, Star2Star mainly serves large businesses in the Southeast region.

Star2Star provides practically every communication feature imaginable. From SIP trunking to Unified Communications (UC) applications, this provider has it covered. They are equipped to provide a number of business phone solutions, including voice calling inside and outside of a company network, mobile network solutions, and UC solutions. Services include conference calling via voice and video, inbound/outbound faxing, presence monitoring, and a multitude of other features. The online portal allows businesses to configure additional features, such as auto attendant, call queue management, call recording, and more. Star2Star also guarantees network uptime over 99%, and can even enable limited-time usage in the event of a power outage.

Star2Star is best suited for mid to enterprise level businesses. For this reason, the company does not provide baseline pricing as it tailors all plans to meet specific business needs. If you would like a quote for your business, contact Star2Star to get in touch with one of their sales consultants.

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Thirdlane Elastic PBX

June 26, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Image

05/05/2016 – New version 7.5.2.1 of Thirdlane Business PBX, Thirdlane Multi Tenant PBX, and Thirdlane Elastic Cloud PBX has been released

Thirdlane Elastic Cloud PBX is a unified communications software platform for large scale hosted Multi Tenant PBX deployments by carriers and Internet Telephony Service Providers. Thirdlane Elastic Cloud PBX offers the same advanced IP PBX functionality as the Thirdlane Multi Tenant PBX – and the next level of scalability and availability. Thirdlane Elastic Cloud PBX integrates Kamailio SIP Server and Asterisk® PBX to provide a highly scalable and reliable Unified Communications software solution for larger service provides.

Try Thirdlane for Free!
Try a demo or download a free trial from the Thirdlane website.

Find Out More
Find out more about Thirdlane products by visiting the Thirdlane website.

Contact Us
For more information, or if you have any questions, please contact us!

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Bicom Systems Call Center PBX

June 21, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Image

Call Center PBX

PBXware Call Center is an IP PBX turnkey Call Center Communications Platform from that is specifically designed to simplify and enhance call management for call centers and contact centers. Call Center PBX enables organizations to effectively start, manage, and grow call campaigns.

Features include:

  • Unlimited ACD queues
  • Unlimited call agents
  • Comprehensive reporting
  • Real-time queue statistics
  • Real-time queue monitoring
  • gloCOM softphone
  • Predictive inbound and outbound dialing
  • Skills based routing
  • …and more!

Learn more on our Call Center PBX webpage.

Or visit our blog to read about the benefits of Hosted VoIP for Call Centers and how to Increase Productivity with Call Center Software

Other Bicom Systems Wikis:

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

GoTrunk Reviews

June 21, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

GoTrunk is a relatively new SIP trunk provider that offers compatibility with most SIP-based PBX systems and easy integration with software like 3CX.

This is advantageous in two ways:
1. Businesses can use any open source SIP-based PBX software.
2. They won’t lose any time or effort spent customizing their SIP solution if they decide to switch SIP trunking providers.

GoTrunk is also relatively easy to setup and install. In fact, the implementation process is just as quick and easy as hosted VoIP’s. If your business already uses IP PBX or a SIP solution, setup can take only a matter of minutes.

GoTrunk SIP telephony solutions include a web dashboard, where businesses can manage their SIP system. The web dashboard can be accessed from any device, anywhere in the world. GoTrunk also provides a variety of tools to help businesses monitor and maintain call quality. Because you can troubleshoot in real-time, you’ll be able to catch and solve issues before they escalate into bigger problems.

Although GoTrunk is a newbie in the SIP trunking world, it’s a sister product of the well-established VoIP provider VoIPstudio.

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Thirdlane Business PBX

June 21, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Image

05/05/2016 – New version 7.5.2.1 of Thirdlane Business PBX, Thirdlane Multi Tenant PBX, and Thirdlane Elastic Cloud PBX has been released

Thirdlane Business PBX is a highly-reliable, cost-effective IP PBX and unified communications software platform that forms the heart of a versatile Voice over IP (VoIP) enabled telephony system. It provides small businesses, larger companies, and multi-site enterprises a flexible combination of the best of open source and commercially developed solutions, offering an alternative to high-cost telephony systems.

In today’s fast-paced business environment, a reliable integrated communications system is critical to success. Thirdlane’s Business PBX delivers by including all the standard telephone system features you expect, plus advanced unified communications capabilities such as integration with email, messaging, and mobile platforms at no extra cost. Thirdlane systems have been field-proven for over ten years across thousands of customers worldwide, and are widely regarded for their unique combination of diverse features, flexibility, and bulletproof reliability.

Thirdlane provides system administrators with all the tools required for easily making changes and adding users or devices. A unique deep customization ability allows each user or group their own feature set to meet virtually every business need. Thirdlane’s Business PBX also boasts an expandable open architecture and an extensive API (application programming interface) to allow integration with third-party applications such as CRM, ERP, accounting systems, and other business tools. The result is that your company can benefit from an easy-to-use yet highly adaptable set of advanced features with a low total cost of ownership.

Key Features:

  • Advanced Call Features: Among the PBX features included are: IVR (auto attendant), conference bridges, call forwarding, transfer, call screening, call parking, call presence, ring groups, hunt groups, find me/follow me, call queues (ACD), direct dial (DID), fax handling, selective call screening and blocking, call recording, intercom, paging, voicemail to email, and much, much more.
  • Auto-Provisioning: Thirdlane Business PBX includes templates for auto-provisioning of devices such as Aastra, Cisco, Linksys, Polycom, Snom and Yealink phones and ATAs. This allows you to easily add phones and devices, individually or in bulk. Entire groups of users can be easily added and provisioned. Templates can be readily customized or added to support new devices.
  • Fine-grained permissions: Allow your administrators to easily control each user’s call permissions and features as appropriate for your business requirements, and to configure dialing rules on a per-user or per-route basis for operational economy and flexibility.
  • Highly Customizable: Choose from a number of Communications Manager and user portal GUI themes, and even customize menus and configuration files. Select from 12 supported languages, or add your own language translations or voice prompts. Easily add custom scripts to support user-requested features and integrate with most third party program’s API.
  • Proven Industry-Standard Components: Thirdlane systems are built with standard, proven components, including the Asterisk® telephony engine and CentOS® Linux, without custom patches. This allows you to easily update them if needed. Thirdlane also supports versions of Digium®-certified Asterisk and Red Hat® Linux for additional peace of mind in critical applications.
  • System Control and Updating: Unlike other solutions, Thirdlane doesn’t deny you root access to control your server for troubleshooting, integrating with third-party components, and performing system maintenance and customization. Software updates are managed by a Thirdlane repository and are easy to install, to keep your system updated with the latest security features. …

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

VoIP Softphones

June 12, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

What is a VoIP Softphone?

A VoIP softphone is a program that installs to and runs from your computer. A VoIP softphone enables you to make calls with just your computer using a VoIP service. Skype, iChat, and GoogleTalk are some of the more popular services, but there are many different ones available for you to choose from.

A VoIP softphone is accessible wherever you have a computer with you and there are both paid and free VoIP softphones. Companies often offer their own proprietary softphones that are configured to work with their service like Cisco and Counterpath.

The VoIP softphones are designed to be intuitive to use and most resemble an actual phone handset. Or you can choose to have the layout show your contacts if that makes calls easier. You can either click the buttons on the interface to dial or use the number pad on your keyboard.

VoIP softphones are only a program on the computer, so a headset with a microphone or an internal microphone and speakers are also needed to make the calls. Headsets prices start at a very affordable $5-$10 for a standard configuration and can get up in the $100-$200 area for a high-quality wireless headset with a long battery life and interchangeable ear pieces.

Who Can Benefit From VoIP Softphones?

Softphones can be used by anybody with a computer. There are a few types of users who can really benefit from the features of a VoIP softphone:

  • VoIP beginners
  • Heavy travelers
  • Telecommuters
  • Call center employees
  • Small businesses
  • Frequent long-distance callers

VoIP beginners can quickly and cheaply explore how the service works by downloading a VoIP softphone to make free computer-to-computer or computer-to-phone calls. Heavy travelers can avoid racking up large bills on their mobile phones or at hotels by using a low-cost VoIP service with a VoIP softphone. Telecommuters can register a VoIP softphone with their office PBX system to enjoy the same call features available to them at the office while they are on the move. Call center employees and small businesses can save on costs by pairing a VoIP service with a softphone to avoid purchasing and maintaining desk phones. International and long-distance rates are much lower when using a VoIP softphone, so those making regular or frequent calls out-of-state or country can cut some major costs.

VoIP Softphone Features

VoIP softphones offer the same features that traditional phones offer and more:

  • Call forwarding
  • Call conferencing
  • Hold capabilities
  • Call transferring
  • Voicemail
  • Greeting capabilities
  • Text, IM, and video capabilities
  • Echo cancellation to improve sound quality
  • Contact list/address book

Softphones also use less energy than phone and phone system hardware, which saves on costs and is useful for green companies.

VoIP Softphone Protocols

VoIP service uses different protocols to determine how the data is processed and transferred over the network. Your VoIP softphone needs to support the same protocol your VoIP service uses. …

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

REVE Systems

June 8, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Image

REVE Systems started in 2003 with a focused approach to serve the IP based communication industry. A Telecommunication and Software Solution provider, REVE Systems has a wide assortment of products, ranging from backbone infrastructure to peripheral products, including middleware. The company today holds a leadership position in Mobile VoIP, Softswitch & Billing and Bandwidth Optimization solutions.

iTel Mobile Dialer

Image

iTel Mobile Dialer is a SIP Softphone, which allows users to make long distance VoIP calls from mobile phones. Compatible with all standard SIP Softswitch, this mobile dialer has been designed keeping VoIP service providers requirements in mind. Being a service provider, you can run this softphone software under your own brand name as iTel Mobile Dialer is completely customizable.

iTel Switch

Image

iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services. iTel VoIP Softswitch has been designed to meet the highest needs of carriers. This VoIP Softswitch also ensures most reliable and cost effective solution that can help VoIP service providers grow as a giant global carrier in VoIP industry. Multilevel reseller support, easy end user interface, integrated billingintegrated billing, intelligent routing and class 4 & 5 Softswitch features are some of the many unique competencies of iTel Switch.

iTel IM Dialer

Image

iTel IM Dialer through its cutting edge Instant messaging (IM) service allows its users (registered under service providers’ brand) to communicate instantly with others which is available in both android and iOS platforms. After registering and authenticating his/her mobile number, users can instantly exchange messages with others whenever, from wherever. Users can also make audio & video call, share files & location, mobile top up and many more. Utilizing the complete customizability and white label branding facility of iTel IM Dialer, service providers can differentiate them from competitors and eventually expand their business successfully.

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

GoForOpen

June 3, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Image

High quality Application Appliance servers using Open Source software!

Our Products:

GoForOpen PBX Series – PBX50 / PBX125 / PBX500 IP PBX for Small to Medium Business

Ultra compact, feature rich PBX for every-day use. It’s completely fanless and silent.

The GoForOpen PBX Series products are high quality IP Telephony appliances with all the basic features and some advanced features only found in large scale phone/ communication systems, but without the additional cost. The GoForOpen PBX Series products were designed to be highly reliable, flexible and easy to use for cost conscious small to medium sized businesses. Take advantage of all the big business features to impress your customers. Reduce your communications costs with VoIP.

Key Features and Benefits:

1. Models support concurrent calls and many extensions:
PBX50 – 50 concurrent calls and at least 200 extensions.
PBX125 – 125 concurrent calls and at least 500 extensions.
PBX500 – 500 concurrent calls and at least 2000 extensions.
2. Interactive Voice Response (IVR) and auto-attendant ensure all calls are answered 24/7 with the highly flexible IVR and auto-attendant even if no one is free to answer the call.
3. Voicemail – Configure a voicemail box for every extension without the additional fees.
4. Link extensions to any phone anywhere – Route calls to an extension to a cell phone, another office extension, a home phone or anywhere else to never miss a call when away from the desk.
5. Support multiple locations with trunks – Connect all your office locations together with a VoIP trunk to save on long distance charges.
6. Web-based configuration – Leverage all the features with the easy to use web-interface.
7. Conference bridge – More than a 3-way conference call, connect multiple participants to a virtual conference room with a professional IVR to handle in-coming calls from participants. Save on the costs of a conference call service with this free built-in feature.
8. Ring groups and hunt groups – Set up ring and hunt groups so that customers can reach your staff in the shortest amount of time.
9. Totally silent fanless chassis.
10. Power consumption < 10W.

Image

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Asterisk cmd MusicOnHold

June 3, 2016 in SIP Phones by transcom  |  No Comments

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination

Synopsis

Play Music On Hold indefinitely.

Description

MusicOnHold([class])

Plays hold music specified by class. If omitted, the default music source for the channel will be used. If you have configured MusicOnHold in musiconhold.conf it will get played automatically if the extension is put on hold. This command FORCES musiconhold music.

The default MusicOnHold class is set with the SetMusicOnHold command (Deprecated in 1.6).

Example

Extension defined in extensions.conf with “forced” MusicOnHold. Remember to Answer before letting the music pour down the line. Otherwise music on hold will not work correctly.
; Answer required as Music On Hold does not answer the call
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()

It is often useful to turn off music on hold in several situations:

  1. when a particular extension calls (originates)
  2. when connectiong to a particular extension
  3. when traversing a particularly expensive network
  4. when connecting to a conference

(So, how do we do handle each situation?)

You can turn off MOH on a per call by using the SetMusicOnHold command (Deprecated in 1.6).

Add a new class to musiconhold.conf
[none]
mode=files
directory=/dev/null

Create a macro in extensions.conf to turn off MOH
[macro-nomusic]
exten => s,1,NoOp(Turn off MOH for this channel)
exten => s,2,SetMusicOnHold(none)

Now call this macro when you dial an extension
exten => 7020,1,NoOp(Dial -> IAX2/outbound/${EXTEN})
exten => 7020,n,Dial(IAX2/outbound/${EXTEN},,M(nomusic))
exten => 7020,n,Hangup

Asterisk 1.6

MusicOnHold(class[,duration])
Plays hold music specified by class. If class is omitted, the default music source for the channel will be used. Change the default class with Set(CHANNEL(musicclass)=…). If duration is given, hold music will be played specified number of seconds. If duration is ommited, music plays indefinitely.

Returns 0 when done, -1 on hangup.

play-fifo (3rd party addition)

This small C program will create if necessary, open and listen on a fifo for slinear audio and delivers it to STDOUT. If STDOUT is blocking, it discards the data. The idea is that you would use it in a custom class in res_musiconhold. Now you can use whatever means you choose in a seperate process to deliver raw 8khz mono slin to the fifo which will be heard as the music class fifo. An example would be to play your line-in into the fifo and the buffer will not overflow because this program does a poll on the STDOUT and discards STDIN when STDOUT is busy.
You can find it here, it is not part of the Asterisk distribution.

See also

Transcom Group :
Transcom ISP Satcoms Domains Askbill Onbored
Make Free VOIP Calls to any destination