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Hosted PBX

June 15, 2013, 6:08 pm
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Hosted PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

"Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX


There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

More Hosted PBX Service Providers


1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • Actual US CLEC
  • Branded customer portals
  • Multiple locations on one virtual PBX
  • Customized pricing.

AstraQom Canada | AstraQom Canada is an innovative provider of Hosted PBX, VoIP IP Telephony, Hosting, and Web Services for Business.
  • All Packages Customizable
  • White label/turnkey packages available
  • Virtual Numbers, Vanity Numbers and Toll Free Numbers
  • AstraQom can provide unified communications for nearly all levels of business

.e4 Technologies - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.

  • 1 MONTH FREE TRIAL
  • VoIP and PSTN Service
  • PBX and ACD
  • Web, Audio & Video Conferencing
  • Skills Based Routing
  • Private/Encrypted and Public IM
  • Voice and Data Archiving
  • Reporting and Archiving
  • PSTN Gateway
  • CRM Integration
  • Web Contact Center


1aVOIP.com

1aVOIP.com delivers hosted PBX services with SIP& Mail feature sets. ...
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Virtual PBX providers

June 15, 2013, 6:17 pm
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List of Virtual PBX Providers


Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.


AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers.

Contact AVOXI at +1-770-937-9735 to learn more about these products and services.

Craig Communications Unlimited US and Canada calling for only $19.95. Privately Owned VoIP service provider for residential and business users. Enterprise features. No contracts. Free equipment!

Phone2call Offers Virtual Number DIDs with Free Virtual PBX. Know why we are very known for our global leadership.
Features:
    • Receive calls from one or many local numbers
    • Forward calls to any phone number (landline or mobile), a VoIP destination or Google Talk
    • Forward calls to multiple destinations simultaneously or by predefined order
    • Record and playback voice messages to callers
    • Direct calls using extension numbers (IVR)
    • Forward calls to a voice-mail box
    • Record calls and send them to an email address
    • Send voice-mail messages to selected email addresses
    • Receive faxes and forward them to an email address
    • Host conference calls
    • Access dial tone for outbound calls (DISA)
    • Block unwanted callers
    • Notify users of selected events via email or SMS messages, with intelligent variables throughout the call flow
Advantages:
    • Extremely powerful and easy-to-use web application for managing and directing incoming phone calls.
    • Drag-and-drop objects being connected together to control exactly how the call progresses.
    • Flexibility, freedom and control over how calls to virtual numbers DID are handled.
    • Personal, and small and medium business use.
    • NO required documentation and inmediate avalaibility
    • FREE when you get your Virtual Number
    • Multilingual including English, Spanish,...


HostingVikings.com hostingvikings.com - Offers you a tailored PBX solution from the cloud.

  • Flat fee ratings with calls to 70 international destinations included
  • Free inbound calls
  • International DID from more that 50 countries included.

Our service can be delivered from our European or US datacenter. Allowing you to give you the best quality needed. Our sip trunking pricings are phone number and call minutes included. Allowing you not to worry about your phone costs.
You can even choose the type of PBX which you would like to have, starting from a HostingVikings pbx over a 3CX hosted environment to a dedicatel Alcatel-Lucent system hosted from your cloud.
Our international MPLS solutions will provide you the quality you need on the location you want, Guaranteed internet connectivity is a key to stable voice calls.


Foehn Ltd

With over 13 years of successful IP communications deployment we have a Hosted VoIP Telephone solution that offers a range of comprehensive features. ...
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the open source SIP TelePresence system

June 17, 2013, 6:37 am
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We (Doubango Telecom) are proud to announce the beta version of our open source SIP TelePresence system (https://groups.google.com/group/opentelepresence)

This is a short but not exhaustive list of supported features on this
beta version:

  • Powerful MCU (Multipoint Control Unit) for audio and video mixing
  • Stereoscopic (spatial) 3D and stereophonic audio
  • Full (1080p) and Ultra (2160p) HD video up to 120fps
  • Conference recording to a file (containers: .mp4, .avi, .mkv or .webm)
  • Smart adaptive audio and video bandwidth management
  • Congestion control mechanism
  • SIP registrar
  • 4 SIP transports (WebSocket?, TCP, TLS and UDP)
  • SA (direct connection to SIP clients) and AS (behind a server, such
as Asterisk, reSIProcate, openSIPS, Kamailio...) modes
  • Support for any WebRTC-capable browser (WebRTC demo client at
http://conf-call.org/)
  • Mixing different audio and video codecs on a single bridge (h264,
vp8, h263, mp4v-es, theora, opus, g711, speex, g722, gsm, g729, amr, ilbc)
  • Protecting a bridge with PIN code
  • Unlimited number of bridge and participants
  • Connecting any SIP client
  • Easy interconnection with PSTN
  • NAT traversal (Symmetric RTP, RTCP-MUX, ICE, STUN and TURN)
  • RTCP Feedbacks (NACK, PLI, FIR, TMMBN, REMB...) for better video
experience
  • Secure signalling (WSS, TLS) and media (SDES-SRTP and DTLS-SRTP)
  • Continuous presence
  • Smart algorithm to detect speakers and listeners
  • Different video patterns/layouts
  • Multiple operating systems (Linux, OS X, Windows ...)
  • 100% open source and free (no locked features)
  • Full documentation (https://code.google.com/p/telepresence/w/list,
http://conf-call.org/technical-guide.pdf)
  • ...and many others

This short list is a good starting point to help you to understand what
you could expect from our TelePresence system.

Technical guide: http://conf-call.org/technical-guide.pdf
Google code website: https://code.google.com/p/telepresence/
Minimal WebRTC Demo client to test all features: http://conf-call.org/
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Android VOIP

June 18, 2013, 4:45 am
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News


FAQ


VOIP Native Applications for Android

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VoIP product from HyberTone, China

June 18, 2013, 5:53 am
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HyberTone Technology Limited
HyberTone Technology Limited is a new high-tech enterprise, which integrates scientific research, developing, producing, and selling into an organic whole. Since it was esta blished, it has been engaged in the research, development and Production of the communication Products. It has owned abundant experience in the development, Production, and sale of software and hardware. The company relies mainly on the sale of solution schemes and regards the sale of its own Products as the complement. The key Products includes all kinds of broadband IP Telephone Terminal station, broadband IP yelephone gateway, communication protocol stack, enterprise exchange server and charge system.


Contact Method,


Customer Manager: Leo

Email: Leo@hybertone.com
MSN: Leo@hybertone.com
Skype: Leohybertone
Mobile: +86 15118089575
Website: www.hybertone.com

Facebook:://www.facebook.com/HyberTone.VoIP
Youtube: http://www.youtube.com/HyberTone

Address: Room 5B, Aozhihao Intergrated Building, Xinzhou 4th Street, Futian District, Shenzhen, Guangdong, China.


Internal and external antenna GSM Gateway
GS-1/GS-1I1Sim1 FXS portSIP&H.323Support IMEI change, VPN-PPTPQuad band GSM moduleBulk SMSRelay encryption
GoIP 1/GoIP 1I1SimSIP&H.323Support IMEI change, VPN-PPTPQuad band GSM moduleBulk SMSRelay encryptionCan be customized
GoIP 4/GoIP 4I4SimsSIP&H.323Support IMEI change, VPN-PPTPQuad band GSM moduleBulk SMSRelay encryptionCan be customized
GoIP 8/GoIP 8I8SimsSIP&H.323Support IMEI change, VPN-PPTPQuad band GSM moduleBulk SMSRelay encryptionCan be customized
GoIP 16/GoIP 16I16Sims SIP&H.323Support IMEI change, VPN-PPTPQuad band GSM moduleBulk SMSRelay encryptionCan be customized


FXS Gateway
HT-912T1port SIP&H.323Support VPN, FAX,Billing supportRouter functionRelay encryptionCan be customized
HT-922T2ports SIP&H.323Support VPN, FAX,Billing supportRouter functionRelay encryptionCan be customized
HT-842R4ports SIP&H.323Support VPN, FAX,Billing supportRouter functionRelay encryptionCan be customized
HT-8828ports SIP&H. ...
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HyberTone

June 18, 2013, 7:20 am
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HyberTone-VoIP Gateway(FXS:1/2/4/8 ports;FXO:2/4 ports),Terminal Phones
HyberTone is a VoIP hardware manufacturer with a low cost from China. Now, we have 1 to 8 ports VoIP Gateways/ATA, are studying & developing the FXO 2/4 ports and two types of terminal phones as below details. All of our products are based on high peformment ARM9 and DSP solution. We also like to highlight our price which will make you surprised.
As HyberTone Company’s all products are designed by itself and it owns all source codes, all products belonging to HyberTone can be customized.
Especially, our current customers have got a good effect and benefit from our products.

Gateways:
1 FXS Port including PSTN bypass
2 FXS Ports including PSTN bypass
4 FXS Ports
8 FXS Ports

2 FXO Ports
4 FXO ports

All above devices:
1.Includes 2 network ports
2.Have router funtction
3.Built-in two protocols the SIP and H.323
4.Embedded encryption module
5.The Server's encrypt software is free charge

Any further informations,pls visit our web site:www.hybertone.com/en
Or contact me with below direct ways:
Skype: Leohybertone
E-mail: Leo@hybertone.com
Mobile: +86 15118089575

Picture.jpg


GoIP-16.jpg

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New Software Releases

June 18, 2013, 7:48 am
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This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

Page Contents


June 2013



May 2013

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VOIP Service Providers Business

June 18, 2013, 9:34 am
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This is a list of Business VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


VOIP Service providers divided by country/continent:

Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.


List of Service Providers:


Alliance Phones is your business ally, when it comes to business communications. We specialize in reliable Hosted PBX systems.

Here is what you will get if you choose Alliance Phones as a business partner.

  • Unlimited Minutes (US and Canada)
  • Advanced phone features (Follow Me, Call Forward, Auto Attendant)
  • No initial setup fees
  • Easy monthly contract


IdeaCom Mid-America- We at Idea Com have served the communication needs of thousands of customers in the Upper Midwest since 1954 through offering wireless/mobility devices, business phone systems, data storage and security assistance and other communications and technology equipment for businesses. Our average technician has worked at Idea Com for more than 10 years and carries at least 4 certifications. Our service personnel are available 24 hours a day and will be available respond to critical communications failures in 2 hours or less. ...
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Web Hosting

June 18, 2013, 9:38 am
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Please add information about web hosting and web hosting providers and companies to this page.

Web Hosting Providers


Please keep this list in alphabetical order

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VOIP Service Providers Residential

June 18, 2013, 1:40 pm
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VOIP Service Providers Business North America

June 18, 2013, 1:43 pm
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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!

Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

If you like this page, please link to it, so Google and other search engines will consider it more important.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page.

Page Contents

Providers in other countries/continents can be found here:

Canada

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VOIP GSM Gateways

June 19, 2013, 8:59 am
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What's a VoIP GSM Gateway?

A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

Who offers VoIP GSM Gateways?


Page Contents

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Asterisk system vendors INDIA

June 19, 2013, 9:00 am
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This is a list of Asterisk system vendors in India.

AAB Asterisk Consultant India


Services Provide:

  • Predictive Dialer
  • Voice Broadcasting & Fax Blasting
  • Medium to Large Multi-Office Business Telephony Systems
  • Call Centers - Local and International
  • Asterisk Dialers and Bulk Calling Systems
  • International Office Telephony Systems
  • Specialty Calling Systems (Entertainment and Personals)
  • Healthcare Application Integration
  • Customer Relationship Management, CRM Application Integration
  • Distributed Server Architecture and Asterisk Load Balancing
  • SIP Express Router, SER Load Balancing
  • Hospitality Telephony Systems (Hotel PBX Integration)
  • Complete IVR Development
  • Local or Datacenter PBX Customization
  • Wireless Telephony Installations
  • Database Integration and Customization
  • Custom Application Development
  • vtiger vicidial integration

IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.

Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream

Asterisk Hyderabad Ph : 9392335385
Asterisk Bangalore Ph : 9392335385

Asterisk IPPBX India (Hyderabad)

Phone: +91 9392335385

Asterisk Consultant : Subramanyam (subbu)

subbu6699@gmail.com

Grandstream hyderabad Ph : 9392335385
Asterisk Chennai Ph : 9392335385
Asterisk Mumbai Ph : 9392335385
Grandstream india Ph : 9392335385
predictive dialer hyderabad Ph : 9392335385
voip hyderabad Ph : 9392335385

Aabhas Bajpai Asterisk System Vendors India !!


Aabhas Bajpai - Asterisk consultant, India

is Having experience in asterisk and VoiP for more then 6+ years
setup many asterisk installation for Companies, call shops and contact centers
Setup HA distributed asterisk servers with 5000 users, distributed in 18 locations worldwide
have done many Small to medium single and distributed load balanced Vicidial Installations
Of more than 300+, Seats plan done in India. ...
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Microsoft Lync

June 19, 2013, 12:45 pm
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The complexity of a Lync implementation is beyond most small and medium-sized businesses. This is further complicated when an installation requires that Lync be integrated with the telephone system.

Introducing Lync Express


LyncExpress2_Appliances.jpg
LyncExpress_SBC_diagram.jpg


To help businesses to manage this complexity, Sangoma has created Lync Express, the only all-in-one Lync server appliance with a built-in VoIP gateway and Session Border Controller.

Microsoft Lync Solutions and Use Cases

Telephony Support for Office 365
Branch Office Support with Continuous Survivable Branch-Office Appliance
IP PBX Replacement with Lync
Extending Lync to the PBX
Session Border Controller for Microsoft® Lync™
Whitepaper: Effectively Bridging Lync and Legacy Communication Nodes in a Mixed Environment

lync_thumbnail.png

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VoIP Gateways

June 19, 2013, 12:49 pm
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Information about VoIP gateways, including VoIP media gateways, FXO gateways, and other VoIP gateways can be found on this page. If you want to add your company's products, please read the Posting Guidelines for Promoting Products and Services

Media Gateways

Media gateways, also commonly referred to as VoIP gateways are devices which bridge conventional telephone networks and equipment to VoIP telephone networks. A typical media gateway has at least one conventional telephone port and at least one ethernet port.

Page Contents


Analog FXO gateways


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Sangoma

June 19, 2013, 1:00 pm
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logo_h30_site_size.gif


Connect with Sangoma

The communications landscape continues to grow in complexity, with more devices, networks, clouds and systems needing to communicate. Sangoma’s portfolio of products enables Service Providers, Carriers, Enterprises, SMBs and OEMs alike to interconnect all of these devices and leverage their existing infrastructure for maximum financial return, while still delivering the most advanced applications and services from the newest technology available. To deliver the IP Telephony functionality and the efficiencies that you and your customers demand, compatibility and dependability across these connections is not just a goal, it’s a must. Sangoma delivers industry leading value and quality, at price points that will make a difference to your ROI. It’s time you connect with Sangoma.

View Sangoma's Products | Solutions | Support
Compatible with Asterisk, Freeswitch, CallWeaver, Yate, 3CX, pbxnsip and others.


Media Gateways

Session Border Controllers

Microsoft® Lync™ Solutions

Multiplexers
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New Software Releases

June 19, 2013, 6:30 pm
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This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

Page Contents


June 2013



May 2013

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Asterisk config features.conf

June 19, 2013, 9:27 pm
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features.conf

Formerly known as parking.conf , renamed to features.conf as of July 17, 2004.

Configuration of Asterisk call parking

parkext => 700                         ; What ext. to dial to park
parkpos => 701-720                  ; What extensions to park calls on        
context  => parkedcalls              ; Which context parked calls are in, need to INCLUDE this in extensions.conf
parkingtime => 45                       ; Number of seconds a call can be parked for (default is 45 seconds)

; after park times out, call will ring back to the original extension

Note: You must restart * after making changes to features.conf, or reload res_features.so

Added to Asterisk with v1.2 release:

Allow to use something other than *8 to pickup a ringing line in your call group.
This feature can be used via the features.conf file in your asterisk directory:

pickupexten = *8 ; Configure the pickup extension. Default is *8

Some SIP phones and ATA's can interfere with service codes like *8 so you can change the pickupexten to some other value. For example, if your extensions are in the range of 201-299, you may want to pick 200.

;transferdigittimeout => 3      ; Number of seconds to wait between digits when transferring a call
                                ; (default is 3 seconds)
;xfersound = beep               ; to indicate an attended transfer is complete
;xferfailsound = beeperr        ; to indicate a failed transfer
;pickupexten = *8               ; Configure the pickup extension. (default is *8)
;pickupsound = beep             ; to indicate a successful pickup (default: no sound)
;pickupfailsound = beeperr      ; to indicate that the pickup failed (default: no sound)
;featuredigittimeout = 1000     ; Max time (ms) between digits for
                                ; feature activation  (default is 1000 ms)
;atxfernoanswertimeout = 15     ; Timeout for answer on attended transfer default is 15 seconds.
;atxferdropcall = no            ; If someone does an attended transfer, then hangs up before the transferred
                                ; caller is connected, then by default, the system will try to call back the
                                ; person that did the transfer.  If this is set to "yes", the callback will
                                ; not be attempted and the transfer will just fail.
                                ; For atxferdropcall=no to work properly, you also need to
                                ; define ATXFER_NULL_TECH in main/features.c.  The reason the
                                ; code is not enabled by default is spelled out in the comment
                                ; block near the top of main/features.c describing ATXFER_NULL_TECH.
;atxferloopdelay = 10           ; Number of seconds to sleep between retries (if atxferdropcall = no)
;atxfercallbackretries = 2      ; Number of times to attempt to send the call back to the transferer.
                                ; By default, this is 2.

[featuremap]
;blindxfer => #1                ; Blind transfer  (default is #) -- Make sure to set the T and/or t option in the Dial() or Queue() app call!
;disconnect => *0               ; Disconnect  (default is *) -- Make sure to set the H and/or h option in the Dial() or Queue() app call!
;automon => *1                  ; One Touch Record a.k.a. Touch Monitor -- Make sure to set the W and/or w option in the Dial() or Queue() app call!
;atxfer => *2                   ; Attended transfer  -- Make sure to set the T and/or t option in the Dial() or Queue()  app call!
;parkcall => #72                ; Park call (one step parking)  -- Make sure to set the K and/or k option in the Dial() app call!
;automixmon => *3               ; One Touch Record a.k.a. Touch MixMonitor -- Make sure to set the X and/or x option in the Dial() or Queue() app call!


[applicationmap]
; don't use e.g #9 for applicationmap or featuremap unless you have changed 'blindxfer' from # to e.g. #1 !
testfeature => *9,callee,Playback,tt-monkeys   ;Play tt-monkes to callee if *9 was pressed - use 'callee' or 'caller'
testfeature3 => *9,self/caller,Monitor,wav|myoutfile|mb 
; note the use of the pipe char '|' here to separate arguments to monitor application

If you set the variable __TRANSFER_CONTEXT, then that context will be used (note the two leading underscores). ...
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VOIP Consultants

June 20, 2013, 1:21 pm
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Creacode

June 20, 2013, 1:36 pm
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Creacode Ltd.
http://www.creacode.com.tr

Creacode SIP Application Server (SIP AS) is a carrier-grade software solution for reliable and cost-effective control of VoIP calls in the center of IP telecommunication networks.

Real-time call starting, routing, redirecting, cut-off and call holding via high level scripting functions
SIP Registrar ve NAT traversal solution for endpoints
SIP IVR voice and DTMF capturing functionalities with multilanguage support
Real-time secure communication with WEB applications, RADIUS servers or directly any relational databases during life-time of each call session
ANI and DNIS based call routing, number normalization and manipulation
Blacklist / Whitelist control for the calls coming from specified IP range.
Strong scripting capabilities for abstraction of SIP and media signaling, call parameters(SIP header) manipulation, timer setting, custom event handling and more
CDR generation into file or database
New call service activation regardless of interrupting ongoing call services
Voice Recording and Voice Mail support
Periodic e-mail notifications related to current state of a session
WEB and Telnet interfaces for server configuration management, CDR reporting and call statistics monitoring
Detailed system and service logs for troubleshooting

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