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VOIP Consultants

June 8, 2013, 1:01 am
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Asterisk Professionals

June 8, 2013, 5:02 am
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The list of Certified Asterisk professionals around the globe.

Australia


Darley Stephen

  • MOBILE 0401597387
  • E-mail: darsdsd@gmail.com
  • all Asterisk / digium installation & support, small / Large scale deployments, Asterisk configuration, Multiple location deployments
  • Predictive Dialer
  • Call Center deployment
  • Industry based telephony installation
  • Global PBX setup
  • IVR
  • Multi site Maintenance
  • E1/PRI/ISDN setup
  • red5 installation & config
  • flash phone customization

Asia (ASEAN) / Myanmar


Asterisk/FreePBX consultant
Asterisk based IP PBX professional service
      • The first Asterisk Consultancy in Myanmar ***

1. IVR
2. FXO/FXS
3. Dial plan
4. Conference Phone configuration
5. Voice Mail.
6. Android/iphone/Nokia Phones with native SIP support for Asterisk using wifi PBX
7. Linux Server Setup.

Technical Support for products: Avaya Nortel, Grandstream, ATCOM, Openvox, Digium, Xorcom, Sangoma, Cisco

Contact: Nay Myo
Email: naymyowin@hotmail.com
http://linocisco.blogspot.com


Asia / India


Enterux

Unified Communication talk to Enterprise Oracle Apps!!!
  • Predictive Dialer
  • 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
  • Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.
  • Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards
  • Installation / maintenance / configuration of linux systems / servers MP124 VOIP GatWays / Phones / devices.
  • Configure Asterisk with Analog phones , USing MP124 Gate Ways or PAP2T Devices
  • Now make your SAP / Custom Oracle based app talk to Asterisk only available with Enterux,

  • #1 Asterisk Development and Deployment partner from India.

Connect with us and find out how we can revolutionize the way you communicate


Asia / IRAN

Omid Mohajerani

Digium Certified Asterisk Professional in IRAN
Training Asteirsk and Nagios in IRAN
Website: http://www.miniatel.com

Haamed Kouhfallah

ELastix Engineer and trainer
Web site: http://www.voip-iran. ...
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Asterisk Consultants Slovenija

June 8, 2013, 6:34 am
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InterAkt Co. d.o.o., Ljubljana, Slovenija

Software solutions

Download Calls Professional for Asterisk PBX (a CDR overview/accounting software)

We offer:
Expert consulting, setup and maintenance of AsteriskPBX systems

    • Home page:http://www.interakt-co.si/
    • Telephone: PSTN +386.1.4210.526
    • Email:support@interakt-co.si
    • Sip:maingate@interakt-co.si
    • Contact: Joze Zivic

I.T. tim d.o.o.


What we do:
  • IT consulting & solutions

What we are:
  • Digium-Certified Asterisk Professional (dCAP)
  • Microsoft Certified Gold partner
  • CISCO Select Partner
  • Partners with IBM HP, TM, IRONPORT...


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Asterisk GUI

June 11, 2013, 11:39 am
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Graphical user interfaces

There are various attempts to write graphical user or administrator's interfaces to Asterisk.
It is ok to list in more than one category as long as the product truly meets the requirements in more than one area. Please sort by alphabetically and do specify the following items when adding an entry:
  • Date of entry (month & year, e.g. "Nov 05")
  • License(s): GPL, BSD, Commercial, Freeware for < X users
  • Technology used: PHP, Perl, Ruby, Java, Tomcat, Horde etc
  • Database used: MySQL, Postgres, SQLite etc.
  • Asterisk version supported: With or without Realtime (ARA)?

Page Contents

Full Service PBX

Products which come as completely configured systems including hardware, or as a complete bare-metal CD/DVD install.

Released (products that are active and in a final released state)
  • VadaXchange browser based user-friendly call control and management interface developed on top of an Asterisk PABX. About time business PABX become user friendly!
  • Aelintra Telecom SARK UCS: High availability, high performance Asterisk-based PBX with on-board network services, automatic phone provisioning, advanced call recording and true Hot Desk support Commercial
  • ansitel: Easy-to-use IP/ISDN-PBX based on Asterisk, ansitel-webinterface 2.0: Full German-/English-Language, for small and medium-sized enterprises. Commercial. Free Live-CD available: www.asterisk-berlin.de.
  • Aquarius Telecoms' TritonOne Multi-Tenant VoIP PBX Server allows companies to offer VoIP services to residential and corporate clients. Based on proven, reliable HP hardware, we offer ease of use and cost effectiveness.
  • asterCC Commercial provides free and full featured IP PBX features, it is designed for hosted PBX solution, besides common pbx functions, it also provides three level billing, automatic invoice so owner could either run it by himself or provide service for customers. Commercial
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Asterisk fax

June 11, 2013, 11:49 am
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VOIP Service Providers B2B

June 11, 2013, 1:10 pm
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Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:

Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.

Page Contents

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PBXww

June 11, 2013, 2:46 pm
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PBXww™ - Power, Flexibility and Freedom in your Virtual PBX

PBXww™ is a fully-featured virtual PBX and also acts as a powerful tool for managing and directing inbound and outbound phone calls. Our unique user interface is specifically designed for simplicity and ease-of-use, with drag-and-drop objects being logically connected together to define the operation of your cloud phone system.

PBXww™ is applicable for both personal and business use, and configurations cover a wide range of applications from simple home use through to complex, multi-branch voice systems. (see full PBXww features).

Please view the PBXww demo video which shows the functionality and flexibility of the platform: PBXww™ - The Free Virtual PBX

Clearly, this platform provides the power and flexibility to construct unlimited call handling scenarios for both personal and business use. Call flows are assembled in a simple and logical manner, with configurations being immediately active and ready to manage phone calls.

Visit one of our Authorized PBXww™ Service Providers to try the PBXww user interface.

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Free Virtual PBX

June 11, 2013, 2:55 pm
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Please list only FREE VIRTUAL PBX service providers and platforms here

  • PBXww - The Free Virtual PBX Service
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Net2Phone Reviews

June 11, 2013, 10:21 pm
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Reviews for the Business VoIP and SIP provider Net2Phone

Net2Phone, is the largest distributor-based VoIP provider in the world. With a global distribution network serving foreign-born communities, Net2Phone provides industry leading voice termination rates. Net2Phone offers a full suite of high-quality, low-cost, VoIP solutions to residential, corporate, and mobile customers in over 160 countries worldwide.

Net2Phone SIP Trunking is a cloud- based phone solution in which we deliver high quality international calling by bundling a VoIP SIP Trunk into existing IP-PBXs and/or VoIP supported gateways. This enables enterprises to replace their legacy phone switches and merge their communications via a single IP circuit, saving considerable infrastructure costs for businesses.
Net2Phone’s SIP Trunking solution will enhance communication, enrich collaboration, and provide measurable cost efficiencies that will improve the way you do business.

In addition to significant cost savings and leading international rates, Net2Phone’s SIP Trunking solution involves no additional hardware or operational costs. There is no charge per channel and simple pay-as-you-go pricing. With certifications by Avaya, Digium, Elastix, and other PBX providers, Net2Phone’s SIP Trunking is compatible and recommended for use with most IP-enabled PBXs and Gateways.

Additional Features include:
  • “Follow Me” VoIP Anywhere capability
  • Automatic recharging
  • International DIDs (international virtual numbers)
  • Gateways Available for Legacy PBX’s
  • Support for all major gateways (AudioCodes, Quintum Cisco ATA,SNOM, Grandstream, ATCOM, Elastix)
  • Support for both digest and IP authentication registrations
  • Support for both high quality and low bandwidth codecs (G.711, G.729, G.723)
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WombatDialer

June 12, 2013, 12:46 am
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WombatDialer is a platform to provide mass outbound calling on the Asterisk PBX.

This can be used to implement many different services. By offering you a set of ready-to-use components and a monitoring GUI, you can create complex solution in minutes.

Typical usage scenarios include:

  • Queue call-backs: allow callers to leave their numbers instead of waiting on a queue, and then being calls them back
  • Web-based contact-us forms: allow customers to be called back by using a web form
  • Automated surveys / Reverse IVRs: call a list of numbers and gather information that is entered as DTMF
  • Number verification services: automatically check the validity of phone numbers
  • Telecasting of pre-recorded messages: send pre-recorded messages to thousands of recipents quickly and efficiently
  • Automated appointment reminders: remind and take appointments, with our without human interventions
  • Automated subscription expiration: automatically call expiring subscriptions, and have clients renew their subscription
  • Automated quality surveys: run quality/performance interviews and feed them back into the QueueMetrics QA module
  • Generic progressive dialing: keep your outbound agents busy by having a dialer compose numbers, handle retries and pre-qualify leads. WombatDialer includes a powerful preview mode with APIs and a web panel.

WombatDialer can work on pre-defined call lists or can dynamically create them over an API (e.g., dial number X after 10:30 AM). It shares the load on one or more PBX servers and has a flexible rescheduling logic to handle missed calls. It is built to be used with your existing Asterisk PBX and does not require separate servers or a separate set of lines. It can call over VoIP or through the public telephone network. It can also act as a power-dialer for a set of agents logged in in Asterisk.

WombatDialer is built to integrate with your business processes, and can receive calls to be made over HTTP and/or notify an external system in real-time of calls made and results gathered.

WombatDialer works natively with the QueueMetrics Call-Center Monitoring Suite in order to produce state-of-the-art campaign analyses and insight.

WombatDialer is currently available as a commecial product, but it is free to use for smaller systems, telephony hackers and Asterisk enthusiasts.
The current version is 0.7.0 as of June 2013.

See also:


Dispayed {HITCOUNTER} times.

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HyperMedia Systems

June 12, 2013, 1:16 am
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Hypermedia Systems


Hypermedia Systems develops and manufactures scalable, professional GSM Gateways and VoIP GSM Gateways for low to high volumes of traffic, starting from 8 ports, with up to 72 cellular channels/ 288 SIMs. Available interfaces: VoIP, E1/T1-PRI and Analog interfaces, as well as GSM, 3G UMTS, CDMA.

These carrier grade Voice Gateways provide opportunities for Alternative carriers, Call termination vendors, and Service providers, as well as for SMB and corporate enterprises.

Read our Testimonials page.


Visit Hypermedia's NewWebsite

We encourage you to visit and get to know it. Find many interesting new areas, like a new downloads page where you can find White Papers, Data Sheets, and Application Notes and more. See our latest addition: Clips & Presentaions.

GSM Gateways and VoIP Gateways for Call Termination

Compact bi-directional gateways for providing cost-effective termination services to or from GSM, UMTS/3G, CDMA, VoIP, E1/T1-PRI or Analog.

Special Offer till 30/06/2013 - Safe revenues with heavy duty termination system! For more information press here.
      • Applies to HG4032p and HG4016p heavy duty series only
      • SLA required


  • Fast Start Series for Call Termination - Plug 'n Sell VoIP GSM Gateway. An ALL-IN-ONE solution for the termination business with a minimum investment.
  • Professional VoIP GSM Gateway. HG-4000 Series. From 12 to 72 cellular ports.
  • Call Termination Systems for VoIP/3G, GSM & CDMA.
  • Combination UMTS / GSM Gateway - HG-5000 Series. The right opportunity for anyone who plans to provide termination services by using both GSM and UMTS cellular networks. You can have several combinations of technologies in the same unit, and receive the traffic by using Embedded VoIP Gateway or by E1/T1-PRI interfaces.


HG4000_side.jpg



GSM Gateways for Cutting Telecom costs

Hypermedia provides a wide range of Call Cost-Saving solutions ideal for cutting costs on daily communication expenses and overall corporate telephony expenditure, while leveraging existing SMB & Corporate business telephony infrastructure. ...
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New Software Releases

June 12, 2013, 5:47 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

  • 2013-06-12 - AdoreSoftphone v2.2.1 released for Android and iPhone5! Download and test the free version with your existing VoIP Platform.
  • 2013-06-05 - SoliCall released a new software technology for noise reduction. On-line audio samples are available here.


May 2013

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Wildix

June 13, 2013, 2:33 am
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LogoWildix_2011_blu.png



WMS— Wildix Multimedia system developed by Wildix


Main Features

  • IAX2 SIP users and trunks full
  • automatic autoprovisioning for W-AIR, Cat-IQ cordless phones
  • features presence monitoring (DND, CF, TIMETABLES)
  • inter system presence monitoring and IM system configuration (NETWORKING, DHCP SERVER, NTP SERVER)
  • full extensions management
  • presence monitoring of sip trunks
  • backup and restore of all system properties
  • WMS users ACL remote fs support and usb devices for data storage
  • remote cdr configuration and Blues
  • support XML application support for Aastra, Grandstream, Snom, Thomson phones
  • Database engine sqlite3

5Vis0pfjq8jnFzK-1DbbiXkkA1LtSoOViQgbvmkd9Xfo8GE_ssuoe5AWbokl9hgn7oL0Krc4OwpVwPMtc7r_P-w3tNbAOSr4b-xRmID86NvQ4LXFoMQ.png



News

WMS 3.0 http://pbx.wildix.com/en/unified-communications-and-voip-solutions/voip-telephony/wms-3-0/.

Resources




Wildix VoIP


pot pourri.jpg

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Cisco Phones

June 13, 2013, 3:00 pm
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Cisco's IP telephony products offer great flexibility, a large feature set, and stand up well to every day corporate use. That being said, they're also not the cheapest in the field.

Cisco's 7960 phone site

Products



Most Cisco IP Phones come pre-loaded with Cisco's proprietary SCCP protocol, for use with the Cisco CallManager or Cisco CallManager Express software. To gain SIP capabilities, one must have a service contract with Cisco and download/install the SIP Firmware for the phone.

Service Contract


Cisco no longer sells service contracts directly, so you have to order the service contract from a Cisco reseller. CDW and Insight are known to sell maintenance contracts for IP Phones.

If you're just looking to obtain firmware for an IP phone, and you are not located inside the United States, then the minimum should be an $8 agreement for one year (P/N CON-SNT-CP7960), chances are you don't need a three year on-site agreement for well over $100. For customers located inside the United States, it is almost 100% likely you will have to purchase the $75, catch-all service contract. By "catch-all" I mean that it includes "firmware" updates and technical support for one year on any Cisco VOIP phone. Lately, CDW has been cancelling orders for US based customers that try to order the $8 support plan after they figure out that they are not supposed to sell it.

The Cisco part number for the < $10 International contract is CON-SNT-CP7960 for the 7960, and CON-SNT-ATA186 for the ATA-186. Note: While trying to get a service contract some have been told that the part number (for either the ATA 186 or 7960) was CON-SNT-PKG1 which runs $60-80.

It will take one of the largest technology companies on the planet several days to process your request at which point you'll get a contract number via email. After registering your contract number on their service center website the technically advanced system will tell you a person will validate your contract and grant you access to the firmware files you need within 24 hours.

If you buy new (and not from a VOIP provider) you should get at least one free support call. You may be able to convince them to provide you with one free download of the firmware without a service contract.

Even if you do not purchase a separate support contract, any new Cisco phone sold through an authorized partner will come with a basic manufacturer's warranty, entitling you to technical support, and in some cases, software downloads. You will need to be registered on the Cisco.com website; call 1-800-553-2447 with that user ID and your phones serial number and ask to open a new case for assistance downloading software. It might not work for all software/firmware, but so long as your phone's warranty is still active, you can at least get a case opened and see what the engineer can provide you with.

If you are buying these phones from someone who is already using them and has a support contract through Cisco, be aware that you will need them to add your billing address to their support contract before Cisco will even talk to you. Cisco seems to be clamping down on the used-phone market by throwing up all of these roadblocks. ...
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Cisco 8961/9951/9971 IP Phones

June 13, 2013, 3:04 pm
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Asterisk AGI

June 14, 2013, 1:50 am
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Asterisk Gateway Interface (AGI)

The Asterisk Gateway Interface is an interface for adding functionality to Asterisk with many different programming languages. Perl, PHP, C, Pascal, Bourne Shell - it's your choice, really.

  • AGI may control the dial plan, called in extensions.conf.
  • Async AGI Introduced in Asterisk 1.6, allows asynchronous AGI scripting.
  • EAGI gives the application the possibility to access and control the sound channel in addition to interaction with the dial plan.
  • FastAGI can be used to do the processing on a remote machine via a network connection.
  • DeadAGI gives access to a dead channel, after hangup. Deprecated since Asterisk 1.6

Page Contents

Gigahz' fork of this, with online lookup if local fails

Applications

  • AGI, DeadAGI, and EAGI: Applications to use in extensions.conf
  • FastAGI: Allows running of AGI applications on a remote machine (improves performance for Asterisk)

Dialing out

If the AGI application dials outward by executing Dial, the script will suspend contact with the Asterisk server until the Dial exits. If you'd like to have additional control after the Dial completes, be sure to pass the 'g' option to the Dial application. ...
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Asterisk auto-dial out

June 14, 2013, 2:21 am
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Asterisk Call Files


Asterisk call files are structured files which, when moved to the appropriate directory, are able to automatically place calls using Asterisk. Call files are a great way place calls automatically without using more complex Asterisk features like the AGI, AMI, and dialplan, and require very little technical knowledge to use.

Page Contents


The Asterisk dial plan extensions.conf responds to someone calling an extension on a channel. If you want to initiate a call from an external application, there are several ways to do this.

There are basically four ways to initiate outgoing calls in Asterisk
  • Use .call files. A call file is a text file that when placed in the correct directory makes Asterisk make an outgoing call.
  • Use the manager API to activate a call. See Asterisk manager dialout
  • Use the Asterisk CLI originate command
  • FollowMe command of Asterisk 1.4: Since this has the abitility to fork (create multiple calls) it could be 'misused' to initiate outgoing calls.

See also additional Digium documents.

New in Asterisk 1.8: A new application Originate has been introduced, that allows asynchronous call origination from the dialplan. ...
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Asterisk auto-dial out deliver message

June 14, 2013, 2:23 am
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Auto-dial out and deliver a prerecorded message

This example showes how to automatically dial out to a phone number and deliver a specified prerecorded message. The outgoing messages may be rerecorded at any time, and the person answering the call is asked to acknowledge receipt. The trigger for the dial-out could be a cron job, an alarm from an network management system, or other external event.



; This is a sample section from an extensions.conf file showing how to auto-dial an outbound call
; and play a prerecorded message when the call is answered.
; To install this application:
;     Create message directory see [record-outboundmsgs] below for details
;     If needed, modify extension numbers and context names to make compatible with your local dialing plan
;     Add the contexts below to your /etc/extensions.conf file
;     Record messages see [record-outboundmsgs] below for details
;     Create a call file to test
;
; The outbound call can be triggered by a call file or manager commands via a telnet session
;
; Creating a call file to call 7026241632 using context @syndeo in the sip.conf file
; set the caller ID to 7022340175 when placing the call
; A correct Caller ID may be required when using some VOIP to PSTN gateway services.
;
; cat <<EOF > /var/spool/asterisk/tmp01
; Channel: SIP/7026241632@syndeo
; Callerid: 7022340175
; MaxRetries: 5
; RetryTime: 300
; WaitTime: 45
; Context: outboundmsg1
; Extension: s
; Priority: 1
; EOF
; mv /var/spool/asterisk/tmp01 /var/spool/asterisk/outgoing
;
;
; For more information see:
;   http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message
;   http://www.voip-info.org/wiki-Asterisk+auto-dial+out
;   http://www.voip-info.org/wiki-Asterisk+manager+dialout
;



[outboundmsg1]
exten => s,1,Set(TIMEOUT(digit)=5)             ; Set Digit Timeout to 5 seconds
exten => s,2,Set(TIMEOUT(response)=10)         ; Set Response Timeout to 10 seconds
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Background(outboundmsgs/msg1)         ; "play outbound msg"
exten => s,6,Background(outboundmsgs/how_to_ack)   ; "Press 1 to replay or 2 to acknowledge receiving this message"
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
; at this point we could do something like reschedule the call to try again later
; or send an email saying the msg was not received,
; or ...


[outboundmsg2]
exten => s,1,Set(TIMEOUT(digit)=5)             ; Set Digit Timeout to 5 seconds
exten => s,2,Set(TIMEOUT(response)=10)         ; Set Response Timeout to 10 seconds
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Background(outboundmsgs/msg2)         ; "play outbound msg"
exten => s,6,Background(outboundmsgs/how_to_ack)   ; "Press 1 to replay or 2 to acknowledge receiving this message"
exten => 1,1,Goto(s,5)   ; replay message
exten => 2,1,Goto(msgack,s,1) ; acknowledge message
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup
; at this point we could do something like reschedule the call to try again later
; or send an email saying the msg was not received,
; or ... ...
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Virtual PBX

June 15, 2013, 5:51 pm
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Virtual PBX is a budget-friendly form of hosted VoIP (Voice over Internet Protocol) that usually only handles inbound calls. A virtual PBX is typically intended for small business VoIP customers with fewer than 10 employees and low-volume telephone traffic.

Page Contents


What Is Virtual PBX?

A virtual PBX is an economy-class version of hosted PBX. Hosted and virtual PBX systems are business VoIP PBX phone systems that transmit calls over the Internet as data.

A virtual PBX offers inexpensive business VoIP telephone service to small businesses. As with a hosted PBX phone systems, a virtual PBX is owned and maintained off-site by a VoIP service provider. A virtual PBX enables a small business telephone system to access enterprise-level features such as auto attendants and voicemail. With virtual PBX small business telephone systems, small start-ups, mom-and-pop shops, freelancers, and entrepreneurs can all present a professional image to vendors, investors, and customers.

Depending on the service provider, a virtual PBX phone system may require a separate phone service for outbound calls.

Virtual PBX Features


Virtual PBX phone systems offer lower costs and fewer features than hosted PBX phone services. Compared to hosted PBX small business telephone systems, virtual PBX service is limited to the most basic fundamentals of business-oriented call controls. Virtual PBX is geared toward simple inbound call-routing for SoHo offices with few personnel, small budgets, and limited calling needs. As with many hosted PBX calling services, most virtual PBX phone systems do not require a contract or term commitment.

Standard features offered with most virtual PBX plans are:

  • Voicemail
  • Auto attendant
  • Unlimited call handling (no busy signals)
  • Call forwarding

Limitations


Virtual PBX phone systems generally:

  • Handle only inbound calls
  • Offer a limited number of extensions
  • May not include Fax over IP (FoIP) services
  • Include a set amount of free minutes
  • May not offer voicemail-to-email
  • May not include international long-distance coverage
  • May not offer Internet fax service
  • May charge extra for conference calling

The features offered vary by virtual PBX VoIP provider. As VoIP service becomes a more common solution for small business telephone systems, many virtual PBX plan features are incorporating the more advanced features of hosted PBX phone systems. Compare plans and prices to determine the best virtual PBX solution.

Cost

Virtual PBX phone prices depend on a variety of factors, such as the features included. Virtual PBX phone service plans can start as low as $9.95 (Grasshopper) per month.

Virtual PBX Service Providers


Some virtual PBX providers include:
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IP PBX

June 15, 2013, 6:03 pm
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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.

Page Contents


Vendor Information


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

3CX - 3CX VoIP IP PBX
__

Bullroar Telecom - Virtual PBX service from Bullroar is an all-inclusive hosted VoIP solution serving small businesses across the Gulf Coast. For one low monthly price per phone users have full access to a broad portfolio of features, virtually unlimited nationwide calling, and the reliability that comes along with our distributed nationwide network.
__

Diamondcard Hosted PBX - Diamondcard Hosted PBX
Diamondcard Hosted PBX - Diamondcard Hosted PBX Wiki

Lean, mean and easy to use. We do initial configuration for you so its really easy to setup. Low cost and using open Source technology. Get started for as low as $9.60/mth. Reseller and private label program too. +1-303-997-0900 #1.

__

.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. ...
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Quatuor Ébène – Beethoven Around the World: The Complete String Quartets...

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