SIP Phones

H.263-Sorenson

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

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Sorenson Spark is an implementation of H.263 for use in Flash Video and Adobe Flash files. FFmpeg uses FLV1 FourCC and Adobe frame identifiers of 0x21, 0x22 and 0x23.

As Apple began to embrace MPEG-4 and move away from other proprietary codecs, Sorenson Media licensed Sorenson Spark (Sorenson H.263) to Macromedia, which was included with Macromedia Flash MX v6 on March 4, 2002. Sorenson Spark is the required video compression format for Flash Player 6 and 7.

Macromedia later tried to find a better video codec. Starting with Flash Player 8 (released in September 2005), the preferred video codec became VP6. Sorenson Spark can be still used in the Adobe Flash CS4 Professional (2008) for Flash Video files (alongside H.264 and VP6).17 According to Adobe engineer Tinic Uro, Sorenson Spark is an incomplete implementation of H.263. It differs mostly in header structure and ranges of the coefficients.

FFmpeg in 2003 added encoding and decoding support for Sorenson H.263.
Asterisk can support it with the chan_rtmp developped by ulex.

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Otubio

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

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Otubio is an International Calling Service provider. The company is headquartered in the USA and offers Cheap International Calls to various destinations all over the world.

Otubio offers mobile apps for the iPhone and Android devices. Otubio can be used either over WiFi, Cellular Data, and Cellular Networks.

See also

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NocRoom Networks

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

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Founded in 2006, NocROOM is located at one of the largest internet exchange points in the world, facilitating excellent inter-connectivity with Latin America, the Caribbean and Europe. NocRoom is a global provider of optimized network solutions that offer the best possible performance, reliability and availability. Our advanced, IP-based networks deliver data, voice and content to clients and partners world-wide.

NocRoom Tier1 Bandwidth Network
Multiple Tier 1 bandwidth providers call NocRoom home; including AT&T, CenturyLink, Cogent, Comcast Business Class, FiberLight, FPL FiberNet, Internap, Level 3, TW Telecom, Verizon Business and XO. NocRoom facility is also home to several Tier 2 carriers.

We offer 24/7/365 tech support with live reps on site every minute of every day to service the thousands of customers we maintain from all over the world.

Services Include:
Colocation, Dedicated Servers, VOIP
PBX (cloud hosted or self hosted office install)

NocRoom has over 100 Gbps of fiber optic connection to the internet with multiple tier 1 carriers.

NocRoom is privately held, bitcoin accepted business with Multi Locations:
Miami FL
Dallas, TX
Las Vegas, NV
Los Angeles, CA

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DooxSwitch

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

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Class 5 Softswitch with Billing and Routing

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www.dooxswitch.com

DooxSwitch is a software platform allowing for rapid VoIP services roll-out. It contains all necessary elements required in successful implementation of various VoIP services..

  • Screenshots here
  • All functionality covered in the detailed manual
  • Newest info and troubleshooting tips can be found in the wiki

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Chicago VoIP

August 23, 2016 in SIP Phones by transcom  |  No Comments

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This is a list of VoIP providers in Chicago. Chicago VoIP companies typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Chicago to the list below.

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    • AltoTelecom is VoIP company that provides VoIP services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com
    • Live Chat Online Free Tech Support for our Clients www.altotelecom.com
    • Free Signup, No Down Payment, No Contracts, Just prepay and start making calls. www.altotelecom.com
  • .e4 SIP/PBX/API/SMS – .e4’s hosted PBX and SIP products lead the way. Check us out for a 1 month trial.
  • Cloud-based phone system for small business in Chicago, cloud hosted PBX, business VoIP system Vox Connect is a cloud-based phone systems provider serving Chicago’s home, small and medium-sized businesses. Our high-quality cost-effective telecom services are custom-designed to fit your business – and your budget. We’ll work with you to learn about your existing telecom infrastructure, understand your telecom needs and customize the configuration of your cloud-based phone system ( also known as cloud based PBX or cloud hosted PBX ) to meet the specific needs of your business.

Why VoxConnect.com?

  • High Quality VoIP Services. Secure, reliable, carrier-grade facilities serving customers in the US and Canada since 2006. Our state of the art Voice over IP infrastructure is monitored 24/7 by our team of expert engineers, ensuring reliability and uninterrupted operations. With VoxConnect.com, your business will experience higher quality service at a lower price.
  • Customized solutions to perfectly fit your small business needs. Every small business is different. We’ll work with you to understand your telecom needs and customize the configuration of your cloud-based phone system ( also known as cloud based PBX or cloud hosted PBX ) to meet the specific needs of your business.
  • Local setup and support. VoxConnect.com is proud to be a part of Chicago’s small business community. We provide on-site network assessment, advise and/or implement the necessary changes and install your new VoIP phones. Service installation with no downtime
  • The phone system and phone service is a vital part of any business. We don’t just ship the VoIP equipment to your location. We’ll migrate and install your cloud based phone system and service without any downtime.

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xCALLY

August 23, 2016 in SIP Phones by transcom  |  No Comments

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xCALLY is a professional Contact Center Suite optimized for Asterisk.

xcally3logosmall.png

Developer: Xenialab
Released: 2010-2011
Official Web Site: www.xcally.com
Product video

Features

– Omni Channel engine: voice, sms, chat, email, video, social media and custom channels
– Drag and Drop Visual WEB IVR
– Customizable Real Time Dashboards
– Contact Manager and Customer Journey management
– Outbound Automatic Dialer
– Customizable Reporting
– Visual Scripting tool
– Triggers and Automations
– API for 3rd party integration

Description

xCALLY is a modular architecture developed using the XCALLY MOTION Real time asynchronous technology
The main modules are:
1. XCALLY Motion Bar: an Omni Channel phone bar designed for the customer care agents, developed to manage the agent status with Login/Logout, Pause, Backoffice, Call features, customizable Call Pop-ups, CTI Computer Telephony Integration for external CRM, ERP and general third party applications and API.

xCallyBar.png

2. xCALLY Core module: a proxy server to connect multiple telephone bars along with the connection to Asterisk for call routing
3. xCALLY Reports: a Web Interface to analyze offline and realtime optimized reports for contact centers

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4. xCALLY Manager GUI: a Web Interface to configure the multi-campaigns, multi-slkills based-routing, time intervals, DID, Centralized Address Books, Call Lists, Sounds etc…

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5. xCALLY Automatic Outbound: a module to manage extension lists and outbound algorithm to perfom automatic outbound distribution calls

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Inbound MultiChannels

xCALLY can dispatch inbound services based on:
– VoIP or Traditional DID: Toll-Free numbers or normal extensions routed through SIP Trunks or traditional PSTN lines

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– Web Calls using the Hand4Shake Web Communication widgets

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REVE Systems Reviews

August 9, 2016 in SIP Phones by transcom  |  No Comments

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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.

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IVRPowers

August 4, 2016 in SIP Phones by transcom  |  No Comments

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IVRPowers provides Interactive Voice & Video Technologies for Cloud-based services. Our products VoiceXML IVR / Video RTC provide easy-to-use, omnichannel, real-time communications, browser-driving interfaces, live chat, web collaboration, video calling, mobile app integration. IVRPowers has offices in Europe (Madrid) and USA (Miami).

Specialities:

VideoRTC, WebRTC, RTMP, VoiceXML, IVR, VoIP, Telephony

Business Lines:

Links:

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Fake False Answer Supervision (FAS) service

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

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Warning!
The service described below is what most would consider fraud and is illegal in many countries.
The information is presented only to assist in identifying this type of fraud.

Fake FAS service is a service that allows you to earn extra money by mixing fake FAS calls into your real traffic. The service generates the needed carrier service messages and creates the needed billable airtime of your traffic, thus, simulating calls to numbers that are “out of coverage” and providing false billing.

WHAT IS FAS?
FAS is commonly referred to as False Answer Supervision and for a regular user this phenomenon means incorrect extra billing of calls: the billing starts earlier than the called party actually picks up the phone.
What is Fake FAS service?
Fake FAS service is a service that simulates calls to numbers that are out of network coverage and that provides false billable airtime to the calling party. The service pretends to be a real mobile carrier and plays back real mobile carrier service messages (e.g. “the number you are calling is not reachable at the moment, please, call back later”) while charging for this.

HOW TO IDENTIFY FAS?
Once you notice one of the following, you are sure to experience a FAS problem :

– the billing of the call starts earlier than the called party picks up the phone

– the call is billed even if the called party is said to be out of the coverage (for example, for a mobile phone you get an announcement “we are sorry, but the party you are calling is unavailable at the moment”)

– the call is billed even if the caller gets to a voicemail.

WHY DO WE EXPERIENCE FAS?
FAS normally takes place when there is no synchronization between a VoIP leg and PSTN leg of the call on a VoIP-to-PSTN gateway. When the call reaches the gateway from the VoIP network, the gateway tries to establish a connection with the called number, but due to incorrect configuration it cannot detect the states of the call, which are advertised by the PSTN network (the states are: “called party ringing”, “called party connected”). And thus the gateway forces the “CONNECT” state and it normally happens immediately after the arrival of the call from the VoIP network or a few seconds after that. But the main idea is that the gateway connects the calls (which means starts the part of the call, which is billed) according to its own settings, but not according to the actual state of the call.

WHY FAS IS A “BAD THING”?
FAS is considered to be a negative phenomenon due to two reasons:

1. The calling enduser may notice that he/she is billed for non-connected calls

2. FAS creates a bigger number of short duration calls (in a situation when there is no answer from the called party, the connected duration of the call is very short (as long as the calling party is willing to listen to Ring Back Tones without being answered), which brings down the ACD/ALOC (average call duration / average length of call) (due to a bigger than usual number of short calls) and brings up the ASR (average success rate) (due to a bigger than usual number of connected calls).

HOW TO EARN MONEY USING FAS?
Normally this is a prerogative of the owners of the gateways that connect the VoIP network to PSTN network. The extra penny is earned by charging the calling party a bit more for the calls using the ability to control the “CONNECT” state of the call. But wholesale transit carriers can also take advantage of using FAS phenomenon to increase profit: by using our service it is possible to mix the needed percent of FAS calls into your existing traffic. This will slightly degrade the ACD parameter of your traffic, increase the ASR parameter and will allow you to earn extra money out of almost nowhere.

HOW DOES IT WORK?
You put the FAS server into routing on your switch. The calls that come to the FAS server will be forwarded to an IVR message, related to the dialed number. The message that will be played back will sound something like this: “the subscriber you are calling is out of reach, please, try to call later”. (IMPORTANT NOTE: it is your responsibility to record such a voice message and to upload it to FAS server.) The message should be exactly the same as the caller would get from the real carrier, to whom this dialed number belongs. The call will be billed from the very beginning and this will be the billable time that will generate additional profit. …

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Unicorn

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

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Product Discontinued (2013)

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