Stay tuned for more information about ClueCon 2012!

Try, Buy or Resell CudaTel from Barracuda Networks. The only PBX designed for FreeSWITCH by the actual creators of FreeSWITCH!

Welcome To FreeSWITCH

The World's First Cross-Platform Scalable FREE Multi-Protocol Soft Switch


FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media.  It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools. More

Feed aggregator

voip-info.org

voip-info.org - Mon, 08/08/2011 - 23:22
Welcome to the VOIP Wiki - a reference guide to all things VOIP
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.

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


[edit]NEWS customers
  • 2011-07-26 - Asterisk VQM Maillist Join Asterisk VQM mail list to discuss Asterisk Voice Quality Monitoring solution. Questions, feature requests are welcome!
  • 2011-07-26 - Nerd Vittles introduces Asterisk 10, PBX in a Flash 1.7.5.6.3, Incredible PBX 2.0, and more...
  • 2011-07-26 - New model ATA AG198 with 1 FXS and PSTN passby is launched by ATCOM.<<more
  • 2011-07-23 - ICT Innovations released new version of ICTFAX Version 0.3.2 , a complete open source fax over ip and billing solution .
  • 2011-07-23 -

Flyers Banners Flash Website Perth

voip-info.org - Mon, 08/08/2011 - 19:29

Grandstream GXV3175

voip-info.org - Mon, 08/08/2011 - 17:54
Grandstream GXV3175 7" Touchscreen IP Multimedia Phone


The Grandstream GXV3175 IP Multimedia Phone, ideal as the ultimate desktop IP multimedia phone.The Grandstream GXV3175 IP Multimedia Phone represents the future in personal IP multimedia communication. Stunning video quality, intriguing user interface with delightful details, rich Web applications embodied in a sleek tablet-like design, distinguishes this product as the ultimate desktop multimedia phone.

The Grandstream GXV3175 IP Multimedia Phone sports a large 7" touch screen color LCD, a tiltable mega pixel CMOS camera with razor sharp clarity, dual network ports with integrated PoE, integrated Wi-Fi, comprehensive auxiliary ports and support for advanced video compression standard H.264/H.263/H.263+.

The Grandstream GXV3175 IP Multimedia Phone redefines the desktop communication experience with a raised level of innovation and integration of state-of-art real time video conferencing, personalized rich media presentation delivery, popular Web and social networking applications and advanced business productivity tools.

Features
  • 7" resistive touch screen color LCD (800 x 480), Tiltable 1.3M pixel CMOS camera with privacy shutter
  • Dual switched 10/100Mbps Ethernet ports with integrated PoE, integrated Wi-Fi (802.11 b/g/n)
  • Dual USB ports, SD/MMC/SDHC, headset, stereo audio output, video output
  • High-fidelity, full-duplex speakerphone with high performance acoustic echo canceller
  • H.264/H.263/H.263+ video compression at bit rate of 32kbps - 2Mbps, frame rate of up to 30fps, video resolution of VGA/WQVGA/QVGA (H.264) and CIF/QCIF (H.263/H.263+)
  • G.711 (a/u-law), G.722 (wideband), G.723.1, G 729A/B, GSM-FR, G.726-32, L16-256 voice codec and AAC, MP3, WMA, Real, Ogg-Vorbis audio codec
  • Multi-language
  • Full Web browser, weather, news, stocks, currency, world clock, calendar, games, Google Voice, Internet radio, YouTube, movie trailer, Last.fm, digital photo frame
  • Photo album integration with Yahoo Flickr/Photobucket/Phanfare, Yahoo/MSN/Google IM, Facebook and Twitter, virtual BLF keys, SDK/API, etc.
  • Smart NAT traversal technology to allow zero-configuration plug-and-play
  • Strong security protection based on TLS/SRTP/HTTPS and AES



SIP Accounts
  • 3 individual SIP accounts plus support for the FREE IPVideoTalk
 
Auxiliary Ports
  • SD/MMC/SDHC
  • 2xUSB 2. ...

Asterisk consultants csa

voip-info.org - Mon, 08/08/2011 - 15:10
Page Contents

IP Video Cameras

voip-info.org - Mon, 08/08/2011 - 15:04
IP cameras are Closed-circuit television (CCTV) cameras that use Internet Protocol to transmit image data and control signals over a Fast Ethernet link. As such, IP cameras are also commonly referred to as network cameras. IP cameras are primarily used for surveillance in the same manner as analog closed-circuit television. A number of IP cameras are normally deployed together with a digital video recorder (DVR) or a network video recorder (NVR) to form a video surveillance system.
Cost advantages

Reduced system cost and added functionality due to general-purpose IP networking equipment infrastructure.
Lower cost of cabling in large installations (CAT5e instead of RG-59 coaxial cable).
Reduced space requirements in large (many camera) CCTV setups because video switching and routing is done via computer and does not need physically large and expensive video matrix switchers.

Extensible network infrastructure
Convergence onto new or existing IP cabling infrastructure, including sites with multiple buildings.
Ability to use Power over Ethernet allowing for one cable to handle power and data.
Capability for deploying with a wireless bridge.
Ability to use legacy coaxial cables with appropriate converters.
Ability to use fiber optic links with appropriate twisted-pair to fiber converters.
Transmission of commands for PTZ (pan, tilt, zoom) cameras via a single network cable.
Simple to add one camera at a time to the system.


360 Degree Cameras

The primary components of the hemispheric camera include a fisheye lens, a high-resolution image sensor and image correction software that is integrated into the camera. Using an ultra-wide angle fisheye lens, the camera captures a 360° hemispheric image of the room and projects it onto a high-resolution image sensor.

When ceiling mounted, the image area of the hemispheric camera covers the entire room. The image in the hemisphere is convex, particularly near the image borders. These image sections are corrected for the viewer by the integrated distortion correction software, allowing a view of the scene from the usual perspective. The virtual PTZ feature allows you to enlarge or move image sections within the hemisphere, just like a PTZ camera yet, with Ip Cameras, this is achieved with no moving parts.


VoIP Functionality

Most IP Cameras support the SIP protocol and enable the use of each cameras to be programed as an extension on a pbx. Cameras from Mobotix offer complete 2 way duplex voice with their integrated microphones and speakers. This works to eliminate the redundancy of having a CCTV system and a paging system and combines it into a single package

Popular Brands

Mobotix
Panasonic
Grandstream.


Where to Buy

    • Mobotix IP Cameras Great Pricing and Support on a full line of IP Cameras- 1-800-213-1896.


DID Service Providers

voip-info.org - Mon, 08/08/2011 - 14:08
A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers see Cheapest ATAs and Service
see domestic USA DIDs for carriers

Free Service Providers Only (Free DIDs)
  • Comity Communications USA DID provider available in most states. Also Origination and Termination Services!
  • 44UK - UK Numbers 44UK - UK NUMBERS are the UK's leaders in inbound call-handling. All UK number ranges are available online (Free and pay-for ranges), together with a complete range of call-handling services. Free Forward to VOIP (SIP/IAX). Free Dealer/Reseller Signup - www.44uk.co.uk.
  • 0207 Numbers 0207 DID Numbers – Get 0207 DID Numbers (Central London) and divert / forward the calls to your existing phone, mobile or VoIP trunk/server. Add voice mail, fax to email, virtual switchboard, call hunting. Change diverts anytime.24/7.
  • Actio.pl Polish business class VoIP service provider for enterprise, contact center and call center clients. Free polish DIDs.
  • aql, Free Signup with PSTN 0870 UK number
  • BGOPEN.NET Free Bulgarian geographical DID numbers. Website is in Bulgarian language only.
  • Brazil Free DIDs Free Brazil DIDs. Limit 5 DIDs per IP. Free SIP Forwarding Only. Click on Free DID section on our site to request.
  • Coms.com-Business Phone Systems Free UK geographical DID numbers.
  • Didforsale Free DID for 24 hours. SIP DID, Each DID comes with 20 channels. Can also do failover and load balancing. Can be used for calling card and call centers.
  • DIDLogic.com DID trading platform. SIP/IAX/H323. No fees. FREE London UK numbers. FREE Skype forwarding! PSTN forwarding (carrier direct wholesale pricing), web-based callback, free SIP accounts. Full time https. Asterisk/Elastix/FreePBX/ATAs supported. Caller ID.

Hosted PBX

voip-info.org - Mon, 08/08/2011 - 14:01
IP Centrex 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.


.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
  • SIP Hardware Compatibility
  • CRM Integration
  • Web Contact Center


1aVOIP.com

1aVOIP.com delivers HOSTED PBX services with SIP & Mail feature sets. Set up your PBX environment with Voice, IM and Videos service, and connect to you customers with phone numbers in: ARGENTINA, BRAZIL, CANADA, CHILE, EL SALVADOR, MEXICO, PANAMA, PERU, PUERTO RICO, UNITED STATES, AUSTRIA, BAHRAIN, BELGIUM, BULGARIA, CROATIA, CYPRUS, CZECH REPUBLIC, DENMARK, ESTONIA, FINLAND, FRANCE, GEORGIA, GERMANY, GREECE, HUNGARY, IRELAND, ISRAEL, ITALY, LATVIA, LITHUANIA, LUXEMBOURG, MALTA, NETHERLANDS, NORWAY, POLAND, SLOVAKIA, SLOVENIA, SPAIN, SWEDEN, SWITZERLAND, UNITED KINGDOM, AUSTRALIA, HONG KONG, JAPAN & NEW ZEALAND

BroadTone Networks Hosted PBX

BroadTone Networks provides a professionally managed Hosted PBX system for small and growing businesses. We are professional and reliable and offer and outstanding product with outstanding customer support.

Or pricing starts at $49.95 and then Just $11.95 per-user.

Same day setup and Free Number porting Available.

New York City Presence, can provide On-site Installation for small additional fee.
call today to get a free quote. 212-937-8835
click4pbx.com click4pbx - Provides a unique trixbox hosted solution - No charge per user, seat or extension. Our hosted pbx includes free extension dialing, free on-network calls. choose your own VoIP provider for outgoing and incoming calls. Our hosted pbx is based on trixbox, you get all features that trixbox offers including: IVR, Queues, Call park, fax, voicemail and more, our plan starts at $99/month for unlimited extensions, your dedicated PBX not a shared, 100 Mbps link speed. you don't have to ever worry about bandwith.


ComCanada Communications Inc. - Provider of Hosted PBX and Virtual PBX phone systems in Canada. We offer full inbound and outbound calling, unlimited North America and Europe long distance, as well as all calling features, including auto attendant, queues, etc. starting at only $10.00/mo. ...

IP01

voip-info.org - Mon, 08/08/2011 - 12:33
IP PBX IP01
  • Open Source Asterisk IP PBX
  • One Port Embedded IP–PBX system



The IP01 is a complete Asterisk Appliance with one FXO or FXS module. It is an embedded open source Linux system with built-in SIP/IAX2 proxy server and NAT functions. It provides a solid, uniform platform for traditional PSTN communications as well as VoIP communications
Targeting for SOHO user and SMB market with an easy to use graphical interface, IP01 provides a cost-saving solution on their telecommunication/data needs. With IP01, company with branch offices in different countries can be easily combined together to work like a virtual single office through internet.

Features:
  • Open Source Asterisk IP PBX
  • Asterisk GUI v2.0
  • OSLEC (Open Source Line Echo Canceller)
  • Configurable IVR menu
  • Voice Mail
  • Voicemail to Email
  • Call Forward
  • Call Waiting
  • Call Transfer (Blind Transfer/ Attender Transfer)
  • Conference Room
  • Password Protect for Conference Room
  • Call Pickup/Call Parking
  • Caller ID
  • Follow Me
  • Call Queues
  • Ring Group
  • Music On Hold
  • Call Detail Record
  • Skype for SIP
  • SIP Trunk
  • IAX2 Trunk
  • PSTN Analog Trunk
  • Call Routing
  • Configure via WEB interface
  • Codec: G.711u/a, G.729, GSM, Speex, G.726
  • Full SSH access
  • 50+ available SIP/IAX2 extensions
  • 20 concurrent calls

Specifications
  • Hardware:
    • CPU: 400MHz Blackfin 532 Chip
    • Flash: 256 MB
    • SDRAM: 64MB
    • LEDs x 4
    • Programmable Reset Button

  • Interface:
    • 1 X RJ45 port
    • 1 X Power port
    • 1 X RS232 port
    • 1 X RJ11 port (FXS/FXO interchangeable)

  • Electrical:
    • Power Input:DC 12V/450 mA

  • Environmental:
    • Operation temperature: 0 to 40 C ( 32 to 104F)
    • Storage temperature: -30 to 65 C (-22 to 149F)
    • Humidity: 10 to 90% no dew

Contact ATCOM: ATCOM Technology co., LTD.
Address: A2F , Block 3 ,Huangguan Technology Park , #21 Tairan 9th Rd, Chegongmiao , Futian District , Shenzhen China
Tel: +(86)755-23487618
Fax: +(86)755-23485319
E-mail: sales@atcomemail.com
Website: http://www.atcom.cn

European Distributors: Neotiq Support and expertise in English and French on the various ATCOM products. ...

Atcom

voip-info.org - Mon, 08/08/2011 - 12:31
Leading Manufacturer of VoIP Products in China
For more information, visit http://www.atcom.cn




Asterisk Appliance Analog ports IP PBX IP01 IP01P IP02 IP04 IP08


Digital Ports IP PBX IP-4B



GSM Built-in IP PBX IP-2G4A IP-4G


IP Phone Entry level IP phone

Private Investigators in Manchester

voip-info.org - Mon, 08/08/2011 - 10:36
Process Server Bolton
Process Server Manchester
Private Investigators in Manchester

Asterisk Professionals

voip-info.org - Mon, 08/08/2011 - 09:39
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 / 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

Argentina / Latin America
Railtion Communication over rails.
Web site: http://www.railtion.com.ar/
E-Mail: info@railtion.com.ar
  • Asterisk consultation
  • Asterisk custom development
  • OpenSER consultation and development
  • Post and Pre paid Billing solutions
  • High availability deployment
  • Best practices in all developments
  • I+D
  • "and everything you can think of about voice over IP communications..."


XARA Telecomunications

One-way Audio

voip-info.org - Mon, 08/08/2011 - 08:51
One-way audio is a common VOIP problem. It is one of the most frequent support questions I receive.
There are many possible causes.

Firmware Outdated firmware in routers, VOIP phones, Firewalls, etc. can cause one-way audio. Ensure you have the latest updates for all the devices in the call path.

Configuration Particularly if NAT is involved in the call path, configuration of the various devices may be a problem.
Check to see if all devices are configured appropiately for your envioronment.

Finding the Cause The basic troubleshooting technique is to use a tool like Ethereal to capture SIP and RTP packets at each point in the call path where packets could be lost. Interperting the resulting captured packets requires some familarity with how networking and VOIP work.

For example if the call path is:

VOIP phone/device --<a>-- firewall --<b>-- sip proxy --<c>-- firewall --<d>-- asterisk

Troubleshooting Steps
  • Start capturing at point <a>
  • Make a VOIP call that will have one-way audio
  • Analyze capture
  • If problem found, fix and retest
  • Otherwise move capture point to the next point (a, b, c, d, etc) and start over

If the problem is intermittent, then a long term simultanous capture at multiple points can be used to attempt to capture a comple call with the problem. Most capture tools will let you capture only traffic from selected devices, so the volume of captured information can be kept to a reasonable size. If a back-to-back SIP user agent (for example a Session Border Controller ) is part of of the call path, then it may be necessary to capture all VOIP traffic at some points to ensure catching the wanted call since the IP addresses can change when traversing this device.



Resources

Virtual PBX providers

voip-info.org - Mon, 08/08/2011 - 06:13
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.

List of Virtual PBX Providers
  • 3Modules.com - Asterisk Virtual PBX Systems starting from 89PLN 23EUR per month. Self-managed or fully administered solutions. We implement VoIP solutions both virtually and onsite. We also extend our offer to include: SIP-to-PSTN termination, DID Porting, DID in 44 Polish area codes, 801 1xx xxx Hotlines, studio recordings and an individual approach to every customer. Our staff speak ENGLISH and are glad to help. Phone: +48 32 7 506 609

  • AccessDirect Hosted PBX - You can boost your business to the next level. Your callers will hear a professionally recorded auto attendant greetings and your staff can work from anywhere. The auto attendant features help tie your staff together, using their existing phone numbers and phones. There is no equipment to purchase, no software to maintain.


  • Acepacket - Enterprise and Residential Broadband PBX/VoIP Services support T.38 - Hosted SBC with LCR and integrated billing system and SIP trunking - Contact Center (IVR- VoiceXML /Jabber / Outbound/Inbound Marketing implementation) - Prepaid/Postpaid CDR / Billing
  • ActiveServe PBX Hosting 3CX, Asterisk, Elastix, and Trixbox PBX Hosting. CAT-5 Data Center, Active NAT Assistance™, Fully Managed Cisco Network, Cloud Platform. No hardware or software to purchase or maintain. 24/7/365 Support. Do-It-Yourself or Turn-Key

loft bed

voip-info.org - Mon, 08/08/2011 - 05:29
loft beds
loft bed


STUN

voip-info.org - Mon, 08/08/2011 - 01:37
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

new VOIP services

voip-info.org - Mon, 08/08/2011 - 00:18
Page Contents
VoIP Service Providers
New VoIP Services Please only post your service once. Repeated postings will be removed.
Note: Providers are encouraged to add their own page on this wiki.
Please keep the entries below 85 characters.

Aug
July
  • Call Shop We are leading providers offering call shop solutions. We offer our services to many businesses and residential clients who wish to make international calls. Visit us now to test our services.
  • 2011-07-05 - Speedflow Communications offers CDR comparison service for VoIP providers


June

May

VOIP Service Providers

voip-info.org - Sun, 08/07/2011 - 20:54
For a list of VOIP to PSTN service providers, indexed by country, please see:

PLEASE DO NOT ADD NEW CATEGORIES HERE.

VOIP provider services, exchanges and other business deals belong under VOIP Service Providers B2B

Please keep your entry in ALPHABETICAL ORDER in relation to the other entries in your section.
If you add a new entry, including an 'added on dd/mmm/yy' would make it easier to notice.

Miscellaneous VOIP related services, including peer-to-peer services, are listed below.

Peer to Peer Service
  • 3fon.net SIP service provider from Ukraine. Main services: Free In-Network Calls and Text Messaging, VoIP International A-Z
  • 45meeting provides simple online conference solution for small business.
  • Alcazar Networks Inc. is offering Wholesale Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! We provide high quality, dependable access to over 3100 rate centers and instant access to over 1,200,000 DIDs - T38 - CNAM Publishing / Lookups - E911 Wholesale SIP (Added on 4-13-2011)
  • AllO.MD - VoIP services provider. Connect and call is easy
  • Axvoice - One of the top VoIP Provider in the US and Canada.
  • Actio. ...

IP04 Open Hardware IP-PBX

voip-info.org - Sun, 08/07/2011 - 17:13
The IP04 is an open (free as in speech) four port IP-PBX that retails for around $450. Both the *hardware* and software are open. The goal of this Wiki page is build community generated content for the IP04, for example configuration and How-To's.


Links
  • http://www.staronetel.com/reseller/ Buy your IP04 or IP08 direct from USA.
  • Wangate Italy Shop for IP02/04/08 Based PBX with Wangate commercially supported software. Open source version also available. Europe resellers welcome.
  • Neotiq Shop Buy your open-source PBX IP04, IP08 or BR4 in France through our shop. Support and new service development upon this platform.
  • VoIPtel SE Actively maintained and available with a Support Contract for the entire line of IPxx PBX's manufactured by Atcom Technology
  • VoIPtel CE Latest firmware for the IP01, IP02, IP04, IP08 and IP BRI manufactured by Atcom Technology
  • IP04 and IP08 producer:ATCOM
  • IP04 page on the Free Telephony Project site.
  • BAPS (Blackfin Asterisk Package System) is the latest firmware distribution for the IP04. BAPS supports package based installation of software (like apt-get or rpm), no compiling is required.
  • VoIPtel GUI has been chosen by Atcom Technology, David Rowe and astfin.org as the default GUI for their IP04, IP08 and IP-BRI.
  • The IP04 is similar to the Asterisk Appliance (AA) and several other embedded Asterisk products. Here is a

Grandstream GXP1200

voip-info.org - Sun, 08/07/2011 - 17:08
Grandstream GXP1200 Entry Level 2-line Enterprise IP Phone



The Grandstream GXP1200 IP Phone is a next generation entry level SIP phone that features 2 line appearances, backlit graphic LCD, 3 XML programmable soft-keys and integrated PoE.

The Grandstream GXP1200 IP Phone delivers superior audio quality, comprehensive telephony features, automated provisioning, security protection for privacy and broad interoperability with most 3rd party SIP devices and leading SIP telephony platforms. The Grandstream GXP2100 supports automated provisioning for easy deployment, advanced security protection for privacy.

It is a perfect choice for enterprise users looking for a high quality, feature rich multi-line IP phone with the best values.





Features
  • 128 x 32 pixel graphical LCD with backlight
  • 2 line appearances with dual color LED and 2 independent SIP accounts
  • 3 XML programmable context sensitive soft keys, 3-way conference
  • High fidelity wideband audio, full-duplex speakerphone with advanced acoustic echo cancellation
  • Dual switched auto sensing 10/100Mbps network ports with integrated PoE
  • Automated provisioning for mass deployment, SRTP and TLS (pending) for security protection


Feature Keys
  • 2 line indicators and Mute button with dual color LED
  • 3 XML dynamic context sensitive soft keys
  • 5 navigation/menu/volume keys
  • 8 dedicated function keys for:
    • Hold
    • Speakerphone
    • Send/Redial
    • Mute
    • Transfer
    • Headset
    • Conference
    • Message (with message indicator)


Reseller VoIP

voip-info.org - Sun, 08/07/2011 - 12:02
Want to be a Reseller Voip?


Visit our website here:Reseller Voip

We are the leaders in Reseller Voip Solutions. We offer our services to Call Shops, Internet Cafe's and individuals who wish to make cheap international calls.


We offer you a complete Voip Reseller solution:

Being a reseller or a partner pays so much! You will have the power to provide VoIP Call Shop
services and be able to brand with your own company's image to your customers. It will appear
as if you invested millions of dollars in the newest network infrastructure in every way, shape and
form, your company name and image is the only one they will associate with the services they are
receiving. You will have all the control, when it comes to Owning your own customer base.

How we do it ?

  • Resellers can create callshops
  • Place your own company’s name, logo and contact information on the billing Software, Making it easier for all your customers.
  • Create & manage your own selling rates. Sell at ANY RATE you like - Add your OWN markup
  • All commercial and financial relations with your call shops are performed by you.
  • Our service team has no contact with any of your customers but you.
  • Run traffic and profit Reports.
  • NO Setup Cost.
  • No Monthly fee's or commitments.
  • We manage all the Technical stuff- all you need to do is SELL & Make Money.
  • 100% private label - so your customers never know you are selling our services.
  • We offer you 24/7 suppor.

For more info:

Visit: Reseller Voip
email: sales@call-shop.com
Mobile: 0027839860150