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trunking gateway

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A trunking gateway is an interface between VoIP and PSTN. It is a device whereby the VoIP line and PSTN line are connected so that an end user can use PSTN phones to make a call over VoIP.

Protocols On E1/T1 side:SS7, PRI, QSIG, R2, V5.2; on IP side: SIP, MGCP, H.248

PDH system includes two major communication systems, ITU-T E1 system and ANSI T1 system.
The E1 system is dominant in Europe and some non-Europe countries. The T1 system is dominant in USA, Canada and Japan.
One major difference between them is :
E1 provides 2.048 Mbps bandwidth but T1 provides 1.544 Mbps bandwidth.

Trunking Gateways


Roytel E1/T1 trunking gateway, RT-EIMS2002

EIMS2002.jpg

Description
Roytel RT-EIMS2002 is a digital trunking gateway which connects PSTN/PBX to IP network. It has carrier-class performance and adapts slot insert frame which can be easily configured as requirements. It provides a maximum of 8 E1 ports in one single device.
RT-EIMS2002 offers 5 modes in one single device, 1/2/4/6/8 E1/T1 ports.
Features
● Carrier-class performance
● Optional and extendable interfaces boards
● 1.5U standard rack-mounted
● Web console
● Flexible call routing
● Support interconversion among multi-protocols, E1<-->IP, E1<--->E1, IP<-->IP

Application networking diagram
The trunking gateway is deployed for telecom carriers.
TG2002 for Carrier application.jpg


The trunking gateway is deployed for enterprise.
TG2002 for enterprise application.jpg


Roytel.jpg

Home page:http://www.roytel.com.cn/index.php?lang=en
Headquarters: Shenzhen, China
E-Mail:kc@roytel.com.cn
Skype ID: yong_chen5
Telephone: +86 755 26743712

OpenVox E1/T1 Truck Gateways, DGW-1001R/DGW-1002R/DGW-1004R

Home page:OpenVox
Headquarters: Shenzhen, China
__E-Mail:__sale@openvox.cn
Telephone: +86 755 82535461


See also




VoIP Wholesale

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Wholesale VoIP Market:


There is no doubt today that VoIP is taking over the telecom market, and every month increases penetration into services and industries. Competitive carriers are looking at the numerous ways to make money from this exploding technology, but there's a lingering question as to whether it is profitable to deliver VoIP in a wholesale model? Their customers, typically Service Providers, are looking for their ‘competitive advantage' into this ‘lowest price' race, leveraging within three key alternatives for packet telephony : “build” , “buy” or “rent”. Business aspect, there’s no need to invest tens of millions of dollars in wholesale VoIP to join in. Many Telecom Companies have done the work for you. They offer a complete, turnkey VoIP business service and equipment. Now you can start wholesale VoIP business with virtually no investment and yet reap great dividends.

Wholesale VoIP Resellers:


In today’s world, Service providers seeking to deliver VoIP to as wide a customer base as possible may find that becoming a wholesale VoIP reseller is the way to go. Wholesale VoIP may be sold to both other service providers and to enterprises or residential customers.

Reselling IP telephony as a wholesale VoIP company is becoming an increasingly popular business model. For many companies, becoming a wholesale VoIP provider hits the sweet spot between profit and market control. Any firm with a well-established customer base is a good candidate for reselling wholesale VoIP.
Becoming a wholesale VoIP reseller is not a decision that should be taken lightly. It does, however, offer the potential of being very lucrative if done right.

Wholesale Consumer Demand:


An important characteristic of the industry is the complex segmentation of consumer demand and rapid change in the characteristics that are being demanded, both at the end customer and in the intermediate ones (wholesale customers).
Demand coming from ‘packed customers'? will be significantly different of the conventional telecommunications one, were telephony was the unique service to provide and differentiation was based on tariff-distance paradigm, being today's service offerings closer to data applications rather than telephony. Voice communication (and not old POT telephony) becomes the common feature into several communications applications and devices, but not the unique one.
Messaging, conference, collaboration, web contact centres, etc … requires a common communication format between parties, which is voice, implemented through VoIP technologies. Heterogeneous and rapidly changing customer demands and products are important dynamic influences on the evolving structure of the telecom industry, resulting into a new value-chain.
Telecommunication markets evolution will be driven by ‘packed customers' demand rather than networks, technology or finance, changing many decades rules into this industry.

Finance in Telecommunication Industry:

Finance institutions had been influencing Telecomm Industry since the beginning, due the business itself was characterized by huge investments, big market shares and bigger capitalization, influencing in many cases top management, who addressed their strategy towards ‘stock' opportunities rather long term and solid business models. WorldCom crash has been an example of this ‘financial market' pressure and wrong business management.
Today, the networks has been deployed. New scenario in Telecoms enable new players to deploy services over broadband without proprietary network and this new generation business will not be anymore capital intensive, let's say these will be innovation intensive.

U.S. VoIP Market:

The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants most lacking network facilities enter the market and drive demand for telephony features and applications, the high-tech market research firm says.
As retail VoIP expands, wholesale VoIP will accelerate quickly, says Bryan Van Dussen, In-Stat analyst. The largest segment remains international VoIP, but we expect the market for local services to surge from 12% of all revenues to 27% by 2010.
Recent research by In-Stat found the following:

  • Consumer VoIP adoption will drive wholesale VoIP revenues to $3.8 billion by 2010 from $1.1 billion in 2006.
  • In-Stat finds small businesses are driving the growth of hosted services in the U.S. Hosted VoIP seats in the U.S. ...

IP PBX

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IP PBX is a phone system utilising IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.

Vendor Information


The following is a list of IP PBX vendors, manufacturers, and service providers.

ALLO PSTN/VOIPPBX - for SOHO with 30 IP extension, upto 6 Analog Extension & upto 4 PSTN trunk. Supports 8 party conference room, Voicemail, IVR , FAX to Email & much more.
__

Flyingvoice- Leading Manufacturer of IP Phone, VoIP Adapter, VoIP Wireless Router and IP PBX in China, NO.1 VoIP Router market share in South Korea.

VOIP Billing

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Hosted Billing Services (in Alphabetical Order)


  • Zoom Soft's Easy VoIP Billing Server provides services for VoIP solution from small to cluster solution for handling thousands of concurrent solution. Our VoIPSwitch is class 5 softswitch which is integrated with billingVoIPSwitch. Also integrated with high feature. Our billing server started with only $50 monthly with 24/7 support. Directly order could be possible with our VoIP billing portal system.
  • Incorpus TeleNetworks Incorpus provides Class 5 and Class 6 softswitches on very affordable monthly rental plans. These are carrier grade switches suitable for Big Enterprises as well as small companies and individuals who are trying to build their own voip company. All switches comes with strong firewalls and bandwidth optimizer and the plans start from as low as just 80$ monthly.Please visit our website and have a live chat with our sales team for any guidance you need. email us for more information at sales@incorpus.in or info@incorpus.in
  • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
    • Note:CloudAstrix SPE Module works with FreeSwitch.
  • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
  • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
  • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
  • Adore All-in-One SIP Server and Client v2.2.1 - new released with Class5 features
  • Aradial AAA for Billing Solutions
  • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. Free signup - best rates on market - low payment amounts

Automatic Call Distributor

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Automatic Call Distributors

Automatic Call Distribution or ACD, is a tool commonly used in the telephony industry. ACD systems are commonly found in any office that handles a large volume of inbound calls. The primary purpose of an Automatic Call Distributor is to disperse incoming calls to contact center agents or employees with specific skills.

The ACD system utilizes a rule based routing strategy, based on a set of instructions that dictates how inbound calls are handled and directed. These rules are often simply based on guiding a caller to any agent as fast as possible, but commonly multiple variables are added, all with the end goal of finding out why the customer is calling. Matching and routing literally thousands of calls to the correct agent is a difficult task, and is often done in concert with Interactive Voice Response and Computer Telephony systems. ACD servers can cost anywhere between a few thousand dollars to close to millions of dollars for a very large call center handling thousands of calls per day.


Automatic Call Distributor Vendors

  • 3CLogic Cloud-Based Contact Center Software 3CLogic is a leading provider of cloud contact center solutions based on an innovative approach, designed to deliver modern-day contact center features to meet the challenges of a modern world. With 3CLogic's ACD functionality, you can set, manage, and adjust call priorities to automatically ensure the most urgent inquiries are always answered first.
  • ICTBroadcast Automatic Call Distributor: Is a Unified automatic call distribution software solution from ICT Innovations . Feature- unifed Auto Dialing, Custom IVR Designer ,Survey Campaign , SMS blasting & marketing , Fax blasting , Voice blasting ,AMD supported, Email marketing and appointment reminder solution .
  • Vocalcom Intelligent distribution of calls is something that Vocalcom has been re-inventing for many years, refining and perfecting to ensure the optimum solution to connect customer and agent.
  • Voicent ACD Software is designed to be configurable to the user. We offer default 'round robbin' call distributions, to the more advanced 'rule & skill based' transfers. Voicent is the leading provider of the Managed Call Center Software.
  • Five9 ACD Software is designed so that any business user can configure it, yet it has all the sophisticated routing features any enterprise requires. Five9 is the leading provider of cloud contact center software.
  • Foehn - We are the experts in IP Communications with over 12 years of successful deployment of Asterisk and open source technology solutions. Our ACDs include skills based routing and sophisticated operator productivity algorithms.
  • AVOXI Provider of ACD /Automatic Call Distribution (ACD) Skills-Based Call Routing.

Sip Trunking Providers

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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specific pages:

logo_gp.png

Zoom Soft is worldwide SIP Trunking Service Provider and Dedicated Server provider. With 24/7 services we always try to give our best services for client. Our services are given below:
  • Cheapest Dedicated Server
  • Easy VoIP Billing Server
    • VoIP Soft Switch
    • SIP Trunking
  • Wholesale A to Z Termination
  • Mobile Dialer
  • DID No
  • SSL Certificates
  • Web Design
  • Web Development
  • Domain Registration
  • Web Hosting

altotelecom-logo.png

ALTOTELECOMVoIP provider for business and Call Centers - AltoTelecom is VoIP company that provides SIP Trunking 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

1comms VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemized per second billing.

1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
  • Actual US CLEC
  • Branded customer portals
  • Multiple geographic locations on one Hosted PBX
  • Coverage in over 50 countries
  • Unlimited inbound on each channel
  • Great for inbound call centers
  • No fee's per user or extension

Incorpus TeleNetworks One stop solution for all your voip needs. DIDs, Tollfree, SIP trunking, WHite label reseller, Private label resellers. Class5 sofswitches, Class 6 softswitches, Carriergrade softswitches all with one time installation and monthly rental plans available. ...

VitalVox

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vitalvox-logo.png

VitalVox is a provider of hosted call center and PBX services. By handling much of the infrastructure, it can provide an interface to world-class call center software while reducing the management costs to clients. Through their PBX services, VitalVox allows businesses to scale quickly without having to invest in and manage telephony switching hardware.

Hosted ACD

VitalVox offers a full-featured Hosted ACD allowing enterprise-grade call centers to use the Cloud as their telephony platform. As a complete call center ACD system, VitalVox provides

  • Skills Based Routing
  • Queue Prioritization
  • Agent Hot-Desking
  • Powerful interactive script builder for agent screens
  • Drag-and-drop call flow builder for IVR menus

Hosted IVR and Visual Call Flow Builder

The Hosted IVR is a powerful tool offered by Vitalvox. It offers a Graphical User Interface (GUI) allowing you to create dialplans, Interactive Voice Response (IVR) systems, and call flows incorporating both telephony functions and call center ACD processes. With it you can interface with internal and external databases and applications. The Dialplan Builder takes the mystery out of building the and inbound service schedules. With its tree-based structure, the Visual Call Flow Builder presents the call flow to you in an easy to read format, while allowing you to rapidly and efficiently trace the structure of the call.

Dialplans are lists of instructions or steps that the call will follow once it reaches the system handling the call. VitalVox offers fully customizable and versatile dialplan capabilities. This gives you full control over the handling and processing of calls. The VitalVox Visual Dialplan Builder enables the user to unleash the combined power of software and telecommunications to easily control and manage calls.

With the Visual Call Flow Builder, you can use your own audio files and external web services to make the hosted IVR truly your own.

Hosted Outbound Dialer

VitalVox offers an self-pacing predictive dialer that is capable of running multiple concurrent outbound dialing campaigns. It is an integral part of our hosted contact center solution. The dialling modes at your disposal are:

  • Preview - the agent is able to review the contact before clicking dial
  • Progressive - the contact is dialed at the same time it is presented to the agent. The agent hears call progress.
  • Predictive dialing - multiple contacts may be dialed per waiting agent. When one connects, it is presented to one of the waiting agents.

All modes of dialling can use the Script Builder technology that allow for sophisticated agent to customer interactions and powerful data collection with adaptive script branching. Our hosted dialer has a complete set of lead management tools for managing your lists. With live dashboards and historical live view of the dialling results, the Outbound Dialer will provide a detailed stats on your agents, list performance and production targets.

Multi Channel Communications

When your business requires more than phones, you'll find our hosted ACD platform handles more than just VoIP traffic.

  • Email
  • Twitter
  • Web chat
and more.

maple4VOIP

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MapleLeaf Technologies

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Astrocom

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www.ecessa.com
Ecessa provides wide area network controller that aggregate mulitple WAN connections with load balancing and automatic failover between WAN connections. Its ClarliLink device includes integrated VoIP features with the ability to load balance SIP connections over multiple WAN links. SIP Proxy, SIP Registration Service and NAT proxy provide the best call integrity - and no dropped calls, even when a link fails.

See Also


VOIP Routers

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There are hundreds of different models of routers available, this is list of routers that may be of interest to VOIP users.

Small Routers with Multiple WAN Interfaces

Having multiple connections to the Internet from different providers can be a challenging configuration puzzle. Some routers are designed to make this easy for simple installations.
  • Astrocom Ecessa ClariLink - multiple WAN interfaces with load balancing and automatic call failover
  • DrayTek multi-WAN routers have special VOIP prioritization features to insure voice goes out first.
  • Linksys RV082 10/100 8-Port VPN Router with 2 WAN interfaces
  • SmartShare Systems FairRouter Dynamic QoS router that automatically detects VoIP traffic (incl. Skype) and ensures high sound quality.
  • SmartNode VoIP routers from PATTON Electronics are industry-renowned for set-it-forget-it reliability. And they come with free support! SmartNode integrates IP and TDM communications for Enterprise and Carrier access networks, offering VoIP gateways combined with IP access routing, WAN transmission, and VoIP border router functionality. SmartNode scales from 1 to 2,048 VoIP or fax calls with various telephony interfaces including analog FXS/FXO and digital ISDN BRI, PRI, DS3 and STM-1.


Small Routers with QOS and built-in VOIP ports

Web Conferencing

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Please add information to this page about Web Conferencing, and web conferencing providers.

Web conferencing is the process of using an Internet connection to conference with one or more people simultaneously through audio or video.

Web Conferencing Providers


Please keep this list of web conferencing providers in alphabetical order

  • Collaboration Solutions Spectranet offering Audio & Video Conferencing solutions and the most innovative Hosted Messaging services.
  • CosmoConf developed in WebRTC & FreeSWITCH is a unified conferencing solution to conduct conferences, meetings, demo, presentations, etc. remotely worldwide.
  • Cisco - Cisco Unified Collaboration Solutions provides all aspects of communication including phone and video conferencing over the web.
  • Drum - Drum provides a web meeting solution utilising WebRTC removing the barriers of software downloads integrated into a website or portal.
  • Google Talk - Google Talk provides web conferencing for free
  • Skype - Skype is the online, free leader of video and web conferencing.
  • PhastCloud - PhastCloud is an easy to use telephony service that uses call-out to quickly conference two or more users. You simply enter your number once and add as many participants as you like. The system then calls them all and places them in your Cloud. voip-info.org users can use trial code V-INFO to sign up for 30 days of unlimited conferencing (US 48).
  • Smart Voice Network AVIDO delivers a truly all-in-one web conferencing solution for all your organization-wide online communication needs. Web conferencing usage can vary widely, depending on business necessities, daily fluctuations in application requirements, and even personal preferences. Using AVIDO as your communication medium offers you easily accessible solutions to any communication challenge – whenever or wherever they arise.
  • Voxeet - Voxeet is a free HD web conferencing software with 3D audio and immersive sound.

See also


Voicepulse

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VoicePulse FIVE


To learn more about VoicePulse FIVE and chat with a Live Representative click here.



VoicePulse FIVE is our fifth generation VoIP platform serving the residential, SMB and Wholesale segments. FIVE is a completely rewritten platform designed from the ground up to provide Internet telephony in the most powerful, flexible and easy to use way possible. Sign up for a free, no obligation evaluation account and be up and running in just minutes.

Company News

2014-11-06 VoicePulse Introduces VoicePulse FIVE, the Next generation in VoIP Services

Pricing

Business Gateway Pricing click here.

Ideal for SMB and Enterprise

Use your existing SIP enabled on-premise softswitch or PBX to make and receive phone calls regardless of call volume.

Channels

Channels are included with your Business Gateway at no additional cost

Endpoints

Endpoints are included with your Business Gateway at no additional cost

Phone Numbers

Choose your number from anywhere in the US or move your existing DIDs to VoicePulse FIVE. There is no fee to port your phone number to VoicePulse FIVE.

Incoming Calls

Incoming calls to a U.S. phone number are $.01 per minute

Incoming Toll-Free Calls

Incoming calls to a U.S. toll-free phone number are $.029 per minute.

Outgoing US & North American Calls

Outgoing calls to the US48 are $.02 per minute

Outgoing International Calls

Competitive International rates.
Look up international termination rates here.


Account Center


  • View your current Statement Balance
  • Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
  • Make instant payments by credit card
  • Add unlimited Channels or call paths to your Trunk
  • View your active phone numbers
  • See our inventory of numbers
  • Instantly activate new numbers
  • Manage E911


Customer Support



Chat with a Live Representative. M-F 9am to 5pm EST


Supported Protocols


  • Session Initiation Protocol (SIP)

Supported Codecs


  • G.729a
  • G.711ulaw
  • G.711alaw
  • GSM
  • ADPCM
  • ILBC

Supported User Agents




  • Asterisk - The Open Source PBX, AsteriskNOW, AA50, SwitchVox
  • Fonality PBXtra
  • trixbox CE, SE, EE, CCE
  • FreeSWITCH
  • Elastix
  • 3CX
  • FreePBX
  • PBX-in-a-Flash
  • OpenSER / Kamailio
  • OpenSIPS
  • Cisco/Linksys SIP devices
  • SIPfoundry
  • Yate
  • Aastra, Grandstream, Snom SIP devices
  • IPitomy
  • OBIHAI technology
  • Kerio Technologies
  • Softphones
  • And more

VOIP Event Calendar

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2015 VOIP Related Event:

June 2015


May 2015



April 2015


March 2015

February 2015

  • 23 OpenSIPS eBootcamp - OpenSIPS online learning program 7 weeks of online classes and labs about SIP and OpenSIPS.
  • 11 -12 SACM 2015 - South Asian Carriers Meet (SACM) is the region’s leading corporate event, attracting the full spectrum of the telecom industry.
  • 9 -10 GCCM London 2015 - Carrier Community is organizing its Annual London 2015 GCCM taken place on 9th & 10th February 2015. Meet 500+ Club Members representing decision-makers from the Tier-1, Tier-2 and Tier-3 from 250+ operators in 40+ countries in London. ...

Asterisk cmd Monitor

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Synopsis

Record a telephone conversation to a sound file

Description

  • Monitor(ext,basename)
  • Monitor(ext,basename,flags) — New feature added to CVS 2004-06-03

The Monitor command starts recording a channel. The channel's input and output voice packets are saved to separate sound files. You may change filenames during a recording by using the ChangeMonitor command. Recording continues until either the StopMonitor command is executed or the channel hangs up.
If you don't specify a full path, the file will be stored in the "monitor" subdir of the path specified with astspooldir in asterisk.conf (so default will be /var/spool/asterisk/monitor).

A more detailed description on recording with Asterisk can be found at Asterisk cmd Record.

Command Parameters

  • ext: The sound file format to save in, which will be also used as the filename extension. Default: wav

  • basename: The base filename to use when saving the sound files. If not supplied, the default basename is constructed on the channel name plus a number, for example, IAX2[foo@bar]-3. The channel's input voice packets will be saved to basename-in.ext and the output voice packets will be saved to basename-out.ext. The default location for saved files is the /var/spool/asterisk/monitor directory.

  • flags:
    • m - New in Asterisk v1.2.0 - If flags contains the letter m, then when recording finishes, Asterisk will execute a unix program to combine the two sound files into a single sound file. By default, Asterisk will execute soxmix and then delete the original two sound files. Note that sox/soxmix may not necessarily understand the sound format (e.g. alaw) and can't therefore mix the in and out files down to one single file. You may specify a different mixing method by setting the MONITOR_EXEC channel variable to the path of the unix program you wish executed, then call Monitor to begin recording. At the completion of recording, the specified unix program will be executed with three command-line parameters: the two sound files and the filename where the program should save the combined sound file. In this situation, earlier versions of Asterisk will not delete the two original sound files; it's up to your program to do that if you need/wish to. The "m" flag is settable through the manager interface.
    • b - New in Asterisk v1.2.0 - Don't begin recording unless a call is bridged to another channel.
    • i - New in Asterisk v1.6.0 - Skip recording of input stream (disables m option).
    • o - New in Asterisk v1.6.0 - Skip recording of output stream (disables m option).

Example 1

When a call is sent to extension 2060, recording of the call will begin, and the caller is sent to conference number 1 with the MeetMe command. ...

Canadian Weather

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Canadian Weather for Asterisk users (UPDATED)

First off, let me thank the original creator of this page, mbuckway, for he showed me the way and I simply brought it to 2015 (and Asterisk 11). I, in NO WAY, take full credit for this page. I simply updated it. All thanks go to him!

Many Asterisk installations (Asterisk@Home, TrixBox etc.) come out of the box with *61 setup to generate weather for US cities. This is all fine and good if you live in the US, but what about all of us up here in the Great White North? With the assistance of an older post http://lists.digium.com/pipermail/asterisk-users/2005-September/117619.html, we managed to create a workable script to generate weather from the Environment Canada website.

(NOTE: since the creation of this page, Environment Canda has gone through many changes. Links and the original script that used to work from here did NOT work anymore, hence why I updated this page.)

The steps to add weather to your Asterisk installation are as follows:
1. Save the script into /usr/lib/asterisk (or other suitable location) named "weather"
2. Modify it to produce weather for your Canadian City (see https://weather.gc.ca/forecast/public_bulletins_e.html). We generated a cleaned up text file for it to run text2wave over below.
3. Setup a cron job to run hourly. (while you could set it to run when you type *61, the weather never updates very often and that means a 2-6 sec wait to generate the weather file - so, we opted for an offline approach)
4. Modify extensions_custom.conf to change it to playback the weather file.

ON TO THE NITTY GRITTY!!! :D

1. Use your favorite editor (eg: nano weather) and create the file called "weather" into /usr/lib/asterisk. (for smooth flow of these instructions)

2. Copy and paste the script below and change the city names to your city/town as listed in the text file from Environment Canada at https://weather.gc.ca/forecast/public_bulletins_e.html. Click on your province, then scroll the text file until you see something like this (but with your city as the title):

City of Ottawa
Gatineau
Prescott and Russell
Cornwall - Morrisburg.
Tonight..Mainly cloudy. Low plus 5.
Thursday..A mix of sun and cloud. High 15. UV index 6 or high.
Thursday night..Mainly cloudy. Low plus 5.

The script to paste and modify into the "weather" file is as follows (REMEMBER: change the Ottawa references to your city name as it appears in the bulletin):

#/bin/bash
rm -f /var/lib/asterisk/sounds/weather.ulaw
echo "Ottawa forecast"> /tmp/weather2.$$
wget -O /tmp/weather.$$ -q https://weather.gc.ca/forecast/public_bulletins_e.html?Bulletin=fpcn11.cwto
grep -A3 "City of Ottawa" /tmp/weather.$$ >> /tmp/weather2.$$
echo "end of forecast">> /tmp/weather2.$$
sed "s/\.\./ /" /tmp/weather2.$$ | text2wave -o /var/lib/asterisk/sounds/en/weather.ulaw -otype ulaw
rm -f /tmp/weather.$$ /tmp/weather2.$$

IMPORTANT: Make sure to type "chmod +x weather" to make the script executeable!!!

3. At a shell prompt, type crontab -e and add the following line:
0 * * * * /usr/lib/asterisk/weather

4. Now add the following lines to extensions_custom.conf:
exten => *61,1,Answer
exten => *61,2,Playback(weather) - this is the name of the ulaw file that will be played when *61 is dialled
;exten => *61,2,AGI(weather.agi) - comment out or remove the AGI line should it be present
exten => *61,3,Hangup

Sometime in the next hour, on the hour, it will execute the file and then you have your weather forecast!
(or type ./usr/lib/asterisk/weather to run it immediately)

Cheers!!!

Toll Free Termination Providers

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Toll-free termination providers allow you to terminate toll-free calls from the US and Canada for free. If you have a large volume of calls to toll-free numbers, some providers will pay you for your calls. Carriers who have direct agreements have a higher success in collection from LD carriers who support the RespOrgs CIC.

Without registration required


Tollfreedollars
Tollfreedollars.com is now offering $.002 per minute for your toll free traffic. No monthly minimums.
To receive the highest payout a contract is required.

HyperCube Leading provider of Wholesale Toll free termination with Collection allowance.
  • Free SIP termination to all NA 8YY destinations.
  • Free SIP trunk from any GLOBAL IP.
  • Free Reporting and Stats on all traffic.
  • Contract required for Collection allowance.
  • CLECs are welcome, MECABS available.
  • Tandem Replacement and full compliance.
  • No Dialer Traffic Allowed.
  • Free Toll Free Numbers for Toll free services available as well.

TollFreeProxy.com - Free Toll-Free SIP Termination
  • Free SIP termination to NANPA toll-free destinations ( 1-800, 1-844, 1-855, 1-866, 1-877, 1-888 )
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STUN

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STUN (Simple Traversal of UDP through NATs (Network Address Translation)) is a protocol for assisting devices behind a NAT firewall or router with their packet routing. RFC 5389 redefines the term STUN as 'Session Traversal Utilities for NAT'.

Note: The STUN RFC states: This protocol is not a cure-all for the problems associated with NAT.


  • STUN enables a device to find out its public IP address and the type of NAT service its sitting behind.
  • STUN operates on TCP and UDP port 3478.
  • STUN is not widely supported by VOIP devices yet.
  • STUN may use DNS SRV records to find STUN servers attached to a domain. The service name is _stun._udp or _stun._tcp

Definitions (from the RFC)

  • STUN Client: A STUN client (also just referred to as a client) is an entity that generates STUN requests. A STUN client can execute on an end system, such as a user's PC, or can run in a network element, such as a conferencing server.
  • STUN Server: A STUN Server (also just referred to as a server) is an entity that receives STUN requests, and sends STUN responses. STUN servers are generally attached to the public Internet.

Various types of NAT (still according to the RFC)
  • Full Cone: A full cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address.
  • Restricted Cone: A restricted cone NAT is one where all requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.
  • Port Restricted Cone: A port restricted cone NAT is like a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.
  • Symmetric: A symmetric NAT is one where all requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.



Closing words (also from the obsolete RFC 3489)

14.6 In Closing

The problems with STUN are not design flaws in STUN. The problems in STUN have to do with the lack of standardized behaviors and controls in NATs. The result of this lack of standardization has been a proliferation of devices whose behavior is highly unpredictable, extremely variable, and uncontrollable. STUN does the best it can in such a hostile environment. Ultimately, the solution is to make the environment less hostile, and to introduce controls and standardized behaviors into NAT. However, until such time as that happens, STUN provides a good short term solution given the terrible conditions under which it is forced to operate.





Standard documents

STUN RFC RFC 3489, now obsolete (Oct 2008)
STUN RFC RFC 5389 (Current as per October 2008)

Update to STUN protocol

STUN standard is currently has been rewritten with RFC 5389. ...

Asterisk High Availability Solutions

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This page outlines the various option available to create high availability for a VoIP PBX. Some are generic solutions while others are PBX specific. Some are complete HA solutions while others are half-baked scripts that do some things but not others.

Before you select a HA solution, carefully read this page on creating / selecting a High Availability solution (see Asterisk High Availability Design )

PBX Specific Solutions (Application Level Cluster)

-These solutions provide clusters that are PBX specific. As noted on the Asterisk High Availability Design wiki page, these solutions create clusters at the Application level and are deeply PBX aware, environmentally aware, trunk aware, etc. The benefit of these solutions is that they are easy to install and provide complete clustering (heartbeat, data synchronizations, failure detection, sharing IP, etc.). The downside of these solutions is that they are PBX specific, so if your PBX software (eg: Asterisk, 3CX, FreeSwitch) is not listed below then you can't use these solutions.

  • HAAst (High Availability for Asterisk) from Generation D Systems adds high availability / clustering to any pair of Asterisk servers. The High Availability for Asterisk (HAAst) add-on offers rapid automatic failover of a failed peer, IP sharing, advanced peer health detection, intelligent synchronization of files and databases, etc. HAAst also supports manual promote/demote for maintenance, a command line interface, a telnet interface, a web based interface, and a developer API. Installation is straight forward, with no additional hardware required, no additional or complex heartbeat/cluster/etc software required either. HAAst is available in Free and Commercial editions, in use at call centers, hospitals, and other high-uptime environments. See High Availability Asterisk (HAAst) for more information.


Custom Asterisk with Generic Linux HA (OS Cluster)

These solutions use generic heartbeat/clustering at the OS level, bundled with a custom version of Asterisk (usually Asterisk + proprietary code).. They may use their own variant of Asterisk, custom GUI, etc. As noted on the Asterisk High Availability Design wiki page, these solutions create clusters at the OS level and are not deeply Asterisk aware, environmentally aware, trunk aware, etc. - they are OS clusters. The benefit of these solutions is that they use off-the-shelf pieces to creaet a cluster, the down side is they mirror corrupt data from one peer to the other, they don't detect deepenvironmental failures, and they can be complex to administer or recover (the HA). Some of these are clearly works in progress as well, others are commercial solutions with slick GUI etc.

  • FreePBX HA FreePBX HA is a commercial module that integrates transparently with FreePBX (2.11 and higher) and Asterisk (1.8 and higher). It is tightly coupled with standard Asterisk, and FreePBX HA offers full control over the complete system, from the ground up. Additionaly, hardware (IPMI/DRAC/ILO) and Virtualized (VMware/libvirt) fencing is supported (and recommended!), More information and documentation is available at the FreePBX HA Wiki. FreePBX HA has won several awards, including the prestegious Best Large Enterprise Solution at IT Expo 2014. FreePBX HA is under active development, as part of the FreePBX Ecosystem.

  • SARK-HA from Aelintra Telecom offers High Availability Asterisk out-of-the box. ...

Asterisk High Availability Design

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High Availability (HA) is normally achieved through "clustering" - which means two machines acting as one for a specific purpose. There are many ways to create a cluster, each with its own benefits, risks, costs, and trade-offs. The terms "High Availability" (HA) and "Clustering" can be overused so beware of the hype. Clustering, and HA have specific (and different!) meanings. If you are responsible for creating a high availability cluster for Asterisk, below are the issues and concepts you should be aware of. This page is intended to be a starting point in the design, creation or selection of a High Availability or Clustering solution for Asterisk.

Please do not add specific product names/links to this page, it is intended to be product neutral.

Fault Tolerance vs Clustering vs High Availability

Before you get started, you should know the difference between these three things. They all mean different things, and you need to be careful what you refer to.

Fault Tolerance

When a fault is detected, it repairs itself quickly. This must NOT result in an outage (for example, a failed CPU that requires a reboot), but the machine must continue working without an interruption. This is normally achieved with Redundant power supplies, network cards, and RAID.

Clustering

When something that is NOT fault tolerant breaks, you must be able to continue on, on different hardware. This may be a CPU failing that requires a reboot., or a kernel panic. At this point, another node should immediately start the services that were on the failed node.

High Availability

HA is a concept, rather than a thing. HA refers to the combination of both Fault Tolerance and clustering, as well as network and power design that removes as many single points of failure as possible. You get to imagine running around with an axe, randomly breaking things, and figuring out what to do when that happens.

Co-Dependence and Autonomy

In order to be a true cluster, the machines (or "peers") must share as little as possible. Some HA solutions involve sharing hardware, software, a logical device, etc .The problem with this approach is that you create a single point of failure. For example, if a cluster shares a hardware channel bank (eg: connected to 2 machines via 2 USB cables), then if the channel bank fails the entire cluster fails. ...
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