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voip-info.org

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 {/COMMENT}
 
 !!NEWS
+* 2013-04-04 - [http://www.sipwise.com/news/announcements/spce-v2_8-release/|sip:provider v2.8 LTS], the open source VoIP appliance, is available
+* 2013-04-03 - [http://voice2phone.com/blog/ringcentral-and-auto-dialer-software|Voice2Phone VoIP Auto Dialer works with Ringcental now]
+* 2013-04-03 - [http://vitelity.com|Vitelity] to Participate in [http://www.vitelity.com/blog/2013/04/visit-vitelity-at-abp-tech-roadshow/|ABP Tech Roadshow]
+* 2013-04-03 - [http://pikawarp.org/?p=2223&utm_source=voip-info.org&utm_medium=blog+post&utm_term=voip+firewall&utm_content=voip+firewall&utm_campaign=blogpostvoipfirewall|Commonly Encountered VoIP Attacks] PIKA Technologies
 * 2013-04-02 - [http://www.newfies-dialer.org/press-1-live-transfer-and-dnc-list/ | Newfies-Dialer V2.1 has been released, autodialer platform]. V2.1 includes new features for Press 1 Live Transfer and DNC list.
 * 2013-04-02 - [http://www.transnexus.com/index.php/fbi-issues-warning-over-voip-tdos-attacks|FBI Issues Warning Over VoIP TDoS Attacks]
 * 2013-04-02 - [http://www.2n.cz/en/news-and-events/2n-helios-ip-force-intercom-equipped-with-video-keeps-schoolchildren-safe/|2N Helios IP Force keeps children safe in USA]
UserCommentTimeVersion
4sipwise2013-04-04 07:4516266
voice2phone2013-04-03 23:2316265
kerbo2013-04-03 20:3716264
kerbo2013-04-03 20:3516263
mennam2013-04-03 17:4716262
mennam2013-04-03 17:4216261
mennam2013-04-03 17:4116260

New Software Releases

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 __Your contributions are welcome, ~~red:but please read the ((How to add information to this wiki)) page and the __((Posting Guidelines for Promoting Products and Services|Posting Guidelines))~~__ ~~red:before you post.~~__
 {maketoc}
+
+!! April 2013
+* 2013-04-04 - [http://www.sipwise.com/news/announcements/spce-v2_8-release/ | sip:provider v2.8 LTS], the open source VoIP appliance, is available - ICE support, new termination billing features, enhanced fraud protection
 
 !! March 2013
 * 2013-03-30 - [http://www.myvoipapp.com/download | miniSIPServer] LTS upgraded to V12. A SIP-PBX for Windows and Ubuntu.
UserCommentTimeVersion
4sipwise2013-04-04 07:551567

How to start a VOIP Business

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 If you are considering starting an ITSP, then this paper is a must-read. It draws from years of experience in setting up ITSPs around the world.
 [http://www.bicomsystems.com/~bicomsys/site/how_to_start_an_itsp/|"How to Start an ITSP"]
 
+---
 !!Whitepaper on "How to Become and ITSP":
 So how does one get started in offering such a service? It is easier than you might think!!!
-[http://voipinnovations.com/blog/index.php/how-to-become-an-itsp-internet-telephony-service-provider/| "How to Become an ITSP"]
+~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/how-to-become-an-itsp-internet-telephony-service-provider/| "How to Become an ITSP"]
 
 ---
 !!Whitepaper on "How to Sell Wholesale VoIP Services:
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/how-to-sell-wholesale-voip-services-choosing-your-target-market/ | How to Sell Wholesale VoIP Services] - Part 1: Choosing your Target Market
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/how-to-sell-voip-part-2-choosing-what-voip-services-to-sell/ | How to Sell VoIP] - Part 2: Choosing what VoIP Services to Sell
----
 
 ---
 !!Whitepaper on "How to Select a Wholesale DID Origination Provider:
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/selecting-a-wholesale-did-origination-provider-part-1-footprint-and-features/ | Selecting a Wholesale DID Origination Provider] - Part 1: Footprint and Features
 Whitepaper [http://voipinnovations.com/blog/index.php/selecting-a-wholesale-did-origination-provider-part-2-monthly-recurring-cost-and-non-recurring-costs/ | Selecting a Wholesale DID Origination Provider] - Part 2: Monthly Recurring Cost and Non-Recurring Costs
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/selecting-a-did-origination-provider-part-3-support-number-porting-and-billing/ | Selecting a DID Origination Provider] - Part 3: Support, Number Porting, and Billing
----
 
+---
 !!Whitepaper on "How to Buy Wholesale SIP Termination:
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/how-to-buy-wholesale-sip-termination/ | How to buy Wholesale SIP Termination]
 

@@ -61,6 +61,10 @@

 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/backoffice-support-the-achilles-heel-of-the-voip-industry-part-1/ | BackOffice Support, the Achilles Heel of the VoIP Industry] Part 1: Service Provisioning
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/the-voip-industrys-achilles-heel-backoffice-support-part-2-service-support-2/ | BackOffice Support, the Achilles Heel of the VoIP Industry] Part 2: Service Support
 ~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/part-3-billing-the-achilles-heel-of-the-voip-industry-voip-innovations/ | BackOffice Support, the Achilles Heel of the VoIP Industry] Part 3: Billing Support
+
+---
+!!Whitepaper on "How to Port a Telephone Number:
+~~orange:Whitepaper~~ [http://voipinnovations.com/blog/index.php/how-to-port-a-telephone-number-voip-innovations/ | How to Port a Telephone Number]
 ---
 
 Whitepaper [http://speedflow.com/download/wholesale_business_2.pdf|How to Start Wholesale VoIP Business]
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v1opinnovations2013-04-04 12:3667
v1opinnovations2013-04-04 12:3566
v1opinnovations2013-04-04 12:3365

inetwork

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 {attachment id=4115}
-inetwork, a division of [Bandwidth.com], gives people the freedom to launch and scale innovate communication services requiring voice, SMS and 9-1-1. Its network (purpose-built for IP applications), industry-leading portal, and competitive pricing provide customers with an easier way to meet their communications needs. inetwork customers like Google, Skype and Pinger benefit from its customizable solutions, nimble operating environment, and responsive customer support.
 
-Bandwidth launched inetwork in 2011 (following acquisitions of dash Carrier services and Vixxi Solutions and integrating its Flex network), specifically to give forward looking companies like Google, Pinger, Skype and LivingSocial a new alternative to traditional telecom s cumbersome way of doing business. inetwork has emerged as the leading choice for innovative and scalable voice, SMS and 9-1-1 solutions
+~~red:__In January of 2013 Bandwidth retired the inetwork brand, collapsing the company's wholesale operations into the Bandwidth brand.__~~
 
-Parent: bandwith
-Website: [http://www.inetwork.com]
-email: sales@inetwork.com
-Phone: (855) 864 7776, option 3
-Key stats
-*A decade of experience in Voice API
-*A decade of experience in IP-based 9-1-1
-*All IP-network purpose-built for next generation networks
-*6th largest provider of telephone numbers in the U.S.
-*The only tier 1 provider with Voice, SMS and 9-1-1
-*Over 23 million users
-*Over 5.8 billion minutes used
-*Over 14 million SMS-enabled DIDs
-*A CLEC in 49 states
-*Covering nearly 22,000 rate centers in over 200 LATAs
-*2011 Growth vs. 2010; 45% increase revenue increase, 41% increase of numbers in use, 71% increase in minutes served
-
-Services
-*Voice Origination and Termination
-*9-1-1 Call Routing
-*SMS
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sfincher2013-04-04 18:4722
admin2013-04-04 17:2320

Bandwidth.com

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-{attachment id=1386}
+Bandwidth is one of the largest telecom companies in the nation, helping over 500 wholesale customers and thousands of enterprise accounts leverage communications as a tool for innovation. We ve made a name for ourselves through our relentless pursuit on innovation, partnering with customers to innovate and change the face of the telecom industry.
 
-Company Name = Bandwidth.com
-Company Type = Private company
-Foundation = 1999
-Location = Cary, NC
-Key People = Henry Kaestner, Chairman, David Morken President & CEO.
-Products = Business phone service, Hosted IP-PBX, SIP Trunking, SIP Origination/Termination, Toll Free Origination. Business-class Internet connections (DSL, T1, NxT1, DS3, OC3, Ethernet), MPLS, VPN, Colocation, Audio Conferencing, Web Conferencing.
-Homepage = [http://www.bandwidth.com/?campaignId=701300000000JBJ|www.bandwidth.com]
-Email = [mailto:info@bandwidth.com]
-Phone = 1-800-808-5150
+The company was founded in 1999 as a reseller of connectivity solutions. We got into the voice business because we were dissatisfied with the available choices in meeting our customers needs for nimble voice solutions. Google and other like-minded thought leaders took note and our voice business was born.
 
+We helped pioneer the concept of SIP trunking in 2006 and added 9-1-1 into our portfolio with the acquisition of dash Carrier Services in 2010. As a result today, we are the only nationwide provider offering integrated voice, 9-1-1, and SMS solutions, all available through intuitive web-based tools.
 
+We think like a software company but with deep telecom knowledge, providing customers with APIs, portals and web based tools that remove telecom s complexity so they can focus on their end users.
 
-^__Bandwidth.com is a complete business communications provider and offering Hosted VoIP, SIP Trunking, Data, and Managed Services to small and medium businesses in the continental United States__^
+We have nearly 300 employees in our Raleigh headquarters, our Denver office and remote locations throughout the country. We are profitable and privately held.
 
+Website: [http://www.bandwidth.com/w&e]
+email: sales@bandwidth.com
+Phone: 855.725.3702
 
-!===Bandwidth.com services include:===
-** Business Phone Service
-**SIP Trunking
-** Hosted IP-PBX
-** Data (T1, Ethernet, etc.)
-** Web and Audio Conferencing
-** Managed network services
+Key info:
+*Founded in 1999
+*the only nationwide provider offering integrated voice, 9-1-1, and SMS solutions
+*All IP-network purpose-built for next generation networks
+*Voice services legacy began in 2002 (dash acquisition)
+*9-1-1 legacy began in 2004 (dash acquisition)
+*A pioneer of SIP Trunking since 2006
+*A decade of experience in Voice API
+*A CLEC in 49 states, covering over 7000 rate centers
+*One of the nation s largest interconnected VoIP footprint, with more numbers than Centurylink, Windstream, Comcast and Time Warner
+*Approximately 300 employees, HQ in Raleigh, office in Denver
 
-
-!===SIP Trunking===
-
-Bandwidth.com SIP Trunks allow businesses to maximize the return on investments in IP PBX technology. Bandwidth.com SIP Trunks allow companies to keep their existing telephone numbers and are fully 911 and e911 compliant. Bandwidth.com SIP Trunks offer the power of 1, enabling businesses to manage one network connection, receive one bill and engage one point of contact for all your local, long distance and broadband Internet needs.
-[http://www.bandwidth.com/sip-trunking]
-
-Bandwidth.com has also recently launched a series of PBX-specific information pages containing specs and setup information for IP PBX systems.
-[http://www.bandwidth.com/content/voip?page=voip_sip_trunking_vendors]
-
--=Bandwidth.com is the preferred SIP Trunking provider for the following PBX manufacturers:=-
-**Mitel [http://www.bandwidth.com/content/voip/mitel]
-**Fonality [http://www.bandwidth.com/content/voip/fonality]
-**Digium [http://www.bandwidth.com/content/voip/digium]
-**AltiGen [http://www.bandwidth.com/content/voip/altigen]
-**Allworx [http://www.bandwidth.com/content/voip/allworx]
-**3Com [http://www.bandwidth.com/content/voip/3com]
-**ShoreTel [http://www.bandwidth.com/content/voip/shoretel]
-**Trixbox [http://www.bandwidth.com/content/voip/trixbox]
-**Asterisk [http://www.bandwidth.com/content/voip/asterisk]
-**Cisco [http://www.bandwidth.com/content/voip/cisco]
-**Pingtel [http://www.bandwidth.com/content/voip/pingtel]
-**Vonexus [http://www.bandwidth.com/content/voip/vonexus]
-**Objectworld [http://www.bandwidth.com/content/voip/object_world]
-**Vertical [http://www.bandwidth.com/content/voip/vertical]
-
-
-For specific questions about using Bandwidth.com SIP Trunks with your PBX, consult the Bandwidth.com User Forums. For example, to answer a question about configuring your Mitel 3300 for use with Bandwidth.com SIP Trunks, go to the Mitel forum: [http://forum.bandwidth.com/]
-
-!===Hosted VoIP===
-Bandwidth.com Hosted IP-PBX gives businesses of all sizes the most productive calling features and innovative IP-PBX features that were traditionally available to only the largest enterprises. All of this without cumbersome in-house systems or software to buy, manage and maintain.
-[http://www.bandwidth.com/hostedvoip/]
-
-Bandwidth.com Hosted IP-PBX offers business the Class V features of a modern IP PBX, including:
-** Find Me / Follow Me
-** Call Hold / Call Transfer
-** Distinctive Ring
-and many more. For a complete listing, visit [http://www.bandwidth.com/content/voip?page=voip_hosted_ippbx_features]
-
-!===Data Services===
-Bandwidth.com data services are designed specifically for the small-to-medium business. Business Internet service has a lot of moving parts to manage. Bandwidth.com Dedicated Internet Services manage the moving parts for you, making it a Simple, Smart and Secure solution for your business.
-[http://www.bandwidth.com/business-internet-access/]
-
-Bandwidth.com data services offer a unique instant pricing tool, as well as an installation tracker, to manage your new circuit from day one. For more information, visit [http://www.bandwidth.com/content/data?page=dia]
-
-!===Managed Network Services===
-Whether determining how to protect your network from the latest Internet virus or providing 24x7x365 monitoring on your network devices, Bandwidth.com can help you lower your costs all while increasing the security and availability of your network.
-[http://www.bandwidth.com/managed-network-services]
-
-Bandwidth.com Managed Network Services are available in two packages
-** Managed Service [http://www.bandwidth.com/content/managednetwork?page=managed]
- This option is a complete "hands off" approach to network management. Bandwidth.com engineers monitor your critical network infrastructure
- 24x7x365, identifying problems and eliminating them before they can slow down your business, or worse, bring down your network.
-
-** Monitor and Alert [http://www.bandwidth.com/content/managednetwork?page=monitoring]
- This option allows you to relax while Bandwidth.com engineers monitor your netowork 24x7x365, tracking key performance metrics and
- identifying potential problems before they affect your business. When key performance or potential trouble thresholds are reached, you
- receive instant notification, allowing you to deal with the problem proactively.
-
-!===Wholesale Services: SIP Origination===
--=Providing Nationwide:=-
-**Enhanced Local Inbound
-**Local Inbound
-**Toll-Free Inbound
-**Termination
-Providing SIP Origination Services to 93% of US Population
-Local calls converted to IP Sonus Media Gateways
-VoIP packets transported over MPLS-enabled backbone
-VoIP packets terminated to your IP Voice application
-Handoff SIP or TDM to your gateway, PBX, or Softswitch
-Full SBC support including AcmePpacket and SER
-!!Origination Service Benefits:
-* Local origination from 6,300+ rate centers, covering 300+ markets
-* Calls directed to one of 2 million IP to PSTN gateways.
-* Full Local Number Portability (LNP) & RESPORG compliance
-* From 1,000 to 10,000 phone numbers available in each unique rate center
-!!!Enhanced Local Origination (ELS) Service Benefits
-** UI/API Order and Management capabilities ((Bandwidth.com NMS))
-** Large local calling areas for reduces costs
-** E911 support
-** Caller Name (CNAM)
-** Caller ID
-** Directory Assistance
-** Operator Assistance
-** Publishing Information
-** 2 Tier breakdown for additional costs savings
-** Thousands of numbers in inventory
-** Search and selection of Vanity Numbers
-** Full support of customer-ported DIDs
-!!!Local Inbound Origination (LI) Service Benefits
-** UI/API Order and Management capabilities ((Bandwidth.com NMS))
-** 3 Tier breakdown for additional costs savings
-** Thousands of numbers in inventory
-** Full support of customer-ported local numbers
-** Low cost/no feature DIDs ready for your business application
-!!!Toll Free Origination Service Benefits
-** UI/API Order and Management capabilities ((Bandwidth.com NMS))
-** Thousands of numbers in inventory
-** Search and selection of Vanity Numbers
-** Full support of customer-ported toll free numbers (RESPORG)
-** Toll free origination supported under North American Dialing Plan (NADP)
-** SIP Toll Free ORIGINATION service provides customers with the same feature set of traditional toll free but with the added benefit of an IP handoff. We enable our customers to deliver new services that support their
-** NADP which includes Continental U.S, Alaska, Hawaii, Puerto Rico, and Canada
-
-
-
-
-
-!!Tech Specs
-{| border=2
-|
-|-
-! !!
-|-
-
-|-
-| DID Coverage Area
-| 93% US Population
-|-
-|-
-| E911 Compliance
-| Yes
-|-
-|-
-| Audio Codecs
-| G.711 and G.729a
-|-
-|-
-| Signaling Protocol
-| TCP and UDP
-|-
-|-
-| Post Dial Delay
-| < 3 Seconds
-|-
-|-
-| Call success rate
-| 99.99%
-|-
-|}
-
-
-!Bandwidth.com DID Management Tools
-^__Bandwidth.com DID Management Tools: Online User Interface (UI) and Full API Support__^
-Tools to help you manage your business.
-((Bandwidth.com NMS))
-
-See Number Management System Architecture((Bandwidth.com NMS Architecture|Example))
-
-* Over ~~red:150,000~~ numbers in inventory
-* Local calling area display
-* Search availability by address, NPA/NXX, or Rate Center
-* Real-time search, selection, and provisioning
-* E911 address, directory listing, publishing, and CNAM management
-* Real-time turn-ups/turn-downs
-* Consecutive block options
-* End point IP address management in real-time
-* Call Data Records (CDR)
-* Bulk Management tools
-* LNP/RESPORG request and timeline support
-* 24/7/365 Customer Care Trouble Management and online ticketing system
-
-!===Wholesale Services: SIP Termination===
-^Voice Termination for the Industry's Leading Carriers^
-Choosing the right partner to enable and extend your service offerings is critical to your success. Your underlying provider must have the ability to deliver consistent performance and the flexibility required to adapt to the rapid changes produced by a shifting telecommunications landscape. With interconnects to Qwest, Level 3, Global Crossing, Verizon, and Ibasis, customers are assured the highest availability and backed by aggressive industry leading SLAs.
-
-!!!Reliability and performance
-Bandwidth.com provides a SIP Proxy service utilizing each carrier s voice network core. With five geographically diverse locations, service is fully meshed to provide redundancy across all paths. Bandwidth.com targets Switch Availability at 99.999 percent and consistently reaches 100 percent month over month. Call blocking is limited on our network and domestic call completion rates have consistently surpassed industry standards.
-!!!Flexibility and convenience
-Bandwidth.com gives you choices. Whether you prefer a TDM or IP interconnection, we have the services to suit your network's requirements. Bandwidth.com provdies SIP as well as TDM hadoffs to support legacy infrastructure.
-!!!Enhanced routing and rating options
-Bandwidth.com's superior network availability, high call completion and competitive, Local Access and Transport Area (LATA)- and Local Exchange Carrier (LEC)- based tiered pricing plans offer carriers of all sizes exceptional value. And for traffic that originates from your end users as IP, Bandwidth.com is able to take advantage of regulatory structures to give you the benefit of even greater cost-effective termination.
+Services
+*Wholesale Voice Origination, Termination, Toll Free
+*SIP Trunking and UC Trunking (Private Label SIP solution)
+*E9-1-1 services for carriers
+*SMS enabled numbers
 
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sfincher2013-04-04 19:0999

ZX20 Mini IP Phone System

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 [http://www.netcloudgroup.co.uk|Netcloud] UK and European Distributor for [http://www.netcloudgroup.co.uk/products/zycoo-voip-systems|Zycoo IPPBX]
 
 __Resellers:__
+
+[http://www.voipon.co.uk/zycoo-zx20-ip-pbx-c-21_704_705.html|ZYCOO ZX20 IP PBX] and the entire product range of [http://www.voipon.co.uk/zycoo-ip-pbx-c-21_704.html|ZYCOO IP PBX] in stock at [http://www.voipon.co.uk|VoIPon]. Save on all [http://www.voipon.co.uk/ip-pbxs-c-21.html|IP PBXs]
 
 [http://www.ipchitchat.co.uk/|IPChitChat] UK and European Distributor of [http://www.ipchitchat.co.uk/index.php/Zycoo-ZX20|Zycoo ZX20]
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VoIPon2013-04-05 15:4813

Web Hosting

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 *[http://www.blueclone.com| Blueclone Networks] offers affordable [http://www.blueclone.com/best/it-consultant/web-site.html| Website Hosting], [http://www.blueclone.com/best/it-consultant/web-site.html| Website Design], [http://www.blueclone.com/best/it-consultant/virtual-helpdesk.html| Virtual Helpdesk] and many other [http://www.blueclone.com/best/technology-company/company.html| business computer services].
 *[http://www.bmchosting.com|BMC Hosting] - Quality Web hosting provider of USA with different package deals. Visit here http://www.bmchosting.com
 *[http://www.bodhost.com/|bodHOST.com] - Fast, Reliable and secure web hosting service with 24*7 free customer service.
+*[http://www.dooxhost.com/|dooxhost.com] -Reliable hosting located in germany ang prague.
 *[http://www.genetechsolutions.com|Genetech Solutions] - Genetech Solutions provides all your web hosting needs in an affordable price
 *[http://www.host-stage.net|HostStage] - HostStage provides fast, reliable, powerful and [http://www.host-stage.net|low cost web hosting] services starting from $1.95
 *Unlimited Trunks/Ext/root access Asterisk Based for more details see [http://www.iguanahosting.com/internacional/en/servidor-pbx-elastix-voip.php]
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z3d3m0n2013-04-06 13:0836

Asterisk Flite

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 * [http://agi-audiotx.sourceforge.net/|asterisk-agi-audiotx]: AGI extension module that adds commands to allow the transfer of audio files to and from Asterisk via an AGI session
 * [http://asterisk-espeak.sourceforge.net/|asterisk-espeak]: dialplan application providing speech synthesis using the eSpeak text-to-speech engine
 * [http://zaf.github.com/Asterisk-eSpeak/|espeak for asterisk 1.6 1.8 and 10]
-
+* [http://www.dudek.org/blog/263#FliteForAsterisk11 | Flite for Asterisk 11 with support for pronunciation exceptions ]
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dudek2013-04-06 22:5019

AT-820P

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+*~~red:Support 2 SIP~~
+*~~red:128*64 DOTS~~
+*~~red:4 softkeys~~
+*~~red:Support POE~~
+
+[http://www.atcom.cn/pro_jpg/AT82001.jpg|{img src=http://www.atcom.cn/pro_jpg/AT82001.jpg width=160px height=120px}]
+AT820P is a user friendly, two lines IP phone with rich function and crystal voice quality,support multi language. Because of the good compatibility with all kinds of standard SIP softswitch and IP PBX system, it is the best choice of VoIP service based on broadband for SMB market.
+
+
+!!Features:
+**High performance Lantiq solution to support crystal clear voice quality
+**SIP support
+**Back light on LCD display
+**Support hardware HD voice and G.722 codec
+**Support 2 SIP lines
+**Elastix certified
+**Dual RJ45 port (WAN+PC) with Router/Switch function
+**Full duplex speaker with HIFI voice quality
+**Suppport VLAN
+**L2TP VPN without IPSec
+**Wall mounted structure
+**Independent RJ9 headset support
+**Customized right tone for different regions
+**Auto provision of configuration
+**Power Over Ethernet
+
+
+
+!!Specifications:
+*__VoIP__
+**Sip 2.0 (RFC3261) and other related SIP RFC
+**2 SIP Accounts
+**STUN
+**Support SIP domain,SIP authentication(none,basic,MD5),NDS
+**Support in band audio, RFC2833 and SIP info,DTMF transmission way
+**SIP Call Forward,Call transfer,Call hold,Call waiting, 3-way talking,Pickup,Redial,Call Park
+**Support DND(Do Not Disturb),Blacklists,Auto answer
+**Dial-peer calling rule, IP to IP call
+**Phone book with 100 records, 50 dialed\answered call\missed calls
+**Support HTTP,FTP TFTP updating the configuration
+**Support multi language
+**Support SMS
+
+
+
+*__Voice__
+**G.711A/u, G722, G.723, iLBC, G.726-16, G.726-24, G.726-32, G.726-40, G.729 Codec
+**G.168 compliant 96ms echo cancellation
+**Jitter Buffer
+**VAD
+**AEC
+**AGC
+**CNG
+**Support HD voice
+**Support HAC
+
+
+
+*__Networking__
+**LAN/PC Support bridge mode
+**LAN support DHCP,Static and PPPoE
+**Support VLAN DATA VLAN and VOICE VLAN
+**Support L2TP VPN without IPSec
+**Qos support diffserv for SIP signaling and voice stream( RTP&RTCP)
+**Support ToS
+**Support SNTP client
+**Support TLS
+**Support SRTP
+
+
+*__Management__
+**Support different level user management
+**Configuration via web,keyboard
+**Firmware and configuration updating via HTTP,FTP and TFTP
+**Support system log
+**Support auto provision for configuration file upgrade
+
+
+*__Hardware__
+**LCD: 128 64 dot matrix
+**FLASH: 8M
+**RAM: 8 16M
+**LED indicator 1 Status Light,2 line indicator, 1 voicemail indicator, 1 headset indicator, 1 mute indicator, 1 hand free indicator
+**Soft key*4
+**Dual Core CPU
+
+
+
+*__Interfaces__
+**Power: Output Power: 12V, 500mA DC
+**LAN: 1*RJ45
+**PC: 1*RJ45
+**Headset jack :1* RJ9
+**Handset jack :1* RJ9
+
+
+
+*__Compliance Certificates__
+**CE: EN55024 EN55022
+
+!!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: [mailto:sales@atcomemail.com|sales@atcomemail.com]
+Website: [http://www.atcom.cn|http://www.atcom.cn]
+
+!!European Distribution:
+[http://www.voipandgo.be/index.php/brands/Atcom/|VoIPandGo.be] Cheap Prices! Quick delivery by UPS. European Distributor
+[http://www.ipchitchat.co.uk/index.php/Atcom-SIP-IAX-Phones|IPChitChat] UK and European Distributor [http://www.ipchitchat.co.uk/index.php/Atcom-SIP-IAX-Phones|Atcom IP Phones] - [http://www.ipchitchat.co.uk/index.php/Atcom-SIP-IAX-Phones/Atcom-AT610-SIP/IAX-IP-Phone|Atcom AT-610P IAX and SIP IP Phone]
+[http://www.netcloudgroup.co.uk|Atcom UK and European Distributor] Netcloud
+
+
+!!See Also
+* ((ATcom))
+* ((VOIP Phones))
 
UserCommentTimeVersion
limn11162013-04-08 06:552

New Software Releases

$
0
0
Diff from version 1567 to 1568 (current)

@@ -4,6 +4,7 @@

 {maketoc}
 
 !! April 2013
+* 2013-04-08 - [http://www.myvoipapp.com | miniSIPServer] V13.1 released to support [http://goo.gl/n9zRC | SIP over TLS].
 * 2013-04-04 - [http://www.sipwise.com/news/announcements/spce-v2_8-release/ | sip:provider v2.8 LTS], the open source VoIP appliance, is available - ICE support, new termination billing features, enhanced fraud protection
 
 !! March 2013
UserCommentTimeVersion
Initialdp2013-04-08 08:591568

HyperMedia Systems

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Diff from version 42 to 43 (current)

@@ -3,6 +3,7 @@

 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 [http://www.hyperms.com/about/testimonials/testimonials.html|Testimonials] page.
 
 !!Visit Hypermedia's ~~red:New~~ [http://www.hyperms.com|Website]
 We encourage you to visit and get to know it. Find many interesting new areas, like a new downloads page where you can find [http://www.hyperms.com/white-papers.html|White Papers], [http://www.hyperms.com/downloads#Data Sheets|Data Sheets], and [http://www.hyperms.com/downloads#Application Notes|Application Notes] and more. See our latest addition: [http://www.hyperms.com/clips.html|Clips & Presentaions].
UserCommentTimeVersion
proza2013-04-08 10:0543

VOIP GSM Gateways

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Diff from version 1019 to 1020 (current)

@@ -1126,6 +1126,8 @@

 * For Call Termination: __[http://www.hyperms.com/form/2013-VCG-special-offer|Special offer - 40% discount]__
 
 !![http://www.hyperms.com| VoIP GSM Gateway - Fast Start Call Termination Kit]
+
+__Check out our [http://www.hyperms.com/products/gsm-gateways/voip-gsm-gateway.html|Call Termination System] video__
 
 The [http://www.hyperms.com/products/gsm-gateways/voip-gsm-gateway.html|VoIP GSM Gateway - HG-4000] is the __ideal solution for anyone who would like to start a new termination business with a minimum investment: just plug it to your broadband carrier, and start selling traffic!__
 The HG-4000 series enables inbound and outbound VoIP and Cellular calls - all in one compact box.
UserCommentTimeVersion
proza2013-04-08 10:151020

IP PBX

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0
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Diff from version 118 to 119 (current)

@@ -320,3 +320,4 @@

 [http://www.ipchitchat.co.uk|IPChitChat] - IP PBX - [http://www.ipchitchat.co.uk/index.php/Atcom-Asterisk-Appliances|Atcom] - [
 http://www.ipchitchat.co.uk/index.php/Positron-Asterisk-Appliance|Positron IP-PBX]
 [http://www.xmartek.com| XmarteK LLC] - 3CX - DENWA - Elastix - Gateways, Transcoders, GSM, Analog Digital IP Phones
+[http://www.mapleleaf-technologies.eu | MapleLeaf Technologies] - pan-European source of premier hardware telephony technologies and solutions
UserCommentTimeVersion
bruckmann2013-04-09 06:44119

MediaProxy Comparison

$
0
0
Diff from version 14 to 17 (current)

@@ -10,6 +10,8 @@

 | [https://github.com/sipwise/mediaproxy-ng | MediaProxy-ng ] || 2.1/2013-02-11 || Sipwise* || C+kernel || yes || yes || yes || yes || yes || yes
 |-
 | [ http://www.rtpproxy.org | RTPproxy ] || 1.2.1/2009-10-19 || Sippy Software || C || ? || ? || ? || ? || ? || ?
+|-
+| [ https://github.com/lemenkov/erlrtpproxy | erlrtpproxy ] || no stable version || Peter Lemenkov || erlang || ? || erlang feature? || no || yes || ? || yes
 |}
 
 {| border="1" cellpadding="2"

@@ -20,8 +22,13 @@

 | MediaProxy-ng || yes || yes || yes || no || [ http://kamailio.org/docs/modules/devel/modules/rtpproxy.html | rtpproxy ], [ https://github.com/sipwise/kamailio/tree/master/modules/rtpproxy-ng | rtpproxy-ng ] || yes || yes || no
 |-
 | RTPproxy || ? || ? || ? || yes || [ http://kamailio.org/docs/modules/devel/modules/rtpproxy.html | rtpproxy ] || ? || no || ?
+|-
+| erlrtpproxy || no || todo || yes || yes || [ http://kamailio.org/docs/modules/devel/modules/rtpproxy.html | rtpproxy ] || ? || no || ? ||[http://lists.sip-router.org/pipermail/sr-dev/2013-January/017928.html | about]
+
 |}
 
+
+See Also
+((MediaProxy Features))
 
 
-kernel based forwarding - It provides better performance than doing it through the daemon, less CPU overhead and lower jitter.
UserCommentTimeVersion
cervajs2013-04-09 11:2617
cervajs2013-04-09 11:2316
cervajs2013-04-09 11:2015

MediaProxy Features

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0
0
Diff from version 1 to 2 (current)

@@ -1 +1,52 @@

+MediaProxy Feaures
+
+Basic functions
+* Basic offer/answer
+* Call recording
+* Music-on-hold
+* Transcoding
+
+
+Configuration & Management
+* Graceful shutdown
+* Configurable IP and UDP port range
+* Automatic load balancing and redundancy
+* Control socket using IPv4/IPv6 UDP/TCP
+* Setting internal parameters via HTTP-JSON interface.
+* Kamailio-style timeout notifications via IPv4/IPv6 UDP/TCP (configurable)
+* JSON command protocol
+
+
+Media
+* Support for any combination of audio and video streams
+* WebRTC-style RTP/RTCP muxing/demuxing from/to two independent channels (automatically)
+* IPv4/IPv6 bridging for RTP/RTCP
+* Video with transcoding.
+* Transcoding offloading using dedicated hardware
+* kernel based forwarding - It provides better performance than doing it through the daemon, less CPU overhead and lower jitter.
+
+Network
+* Multiple networks bridging (more than just three - internal, external, ipv6).
+* STUN/TURN/ICE
+* SRTP / ZRTP
+
+Accounting and statistics
+* Real-time sessions statistics
+* RADIUS accounting (start / interim update / stop, configurable)
+* JSON stats retrieval via HTTP (queries like http://localhost:8442/json/?callid=0003e30c-c50c0134-763a6f53-359481cc@192.168.0.100&payload=0 - available parameters are - payload, callid, mediaid, tag, remote IP)
+
+{"http_query":{"callid":"0003e30c-c50c0134-763a6f53-359481cc at 192.168.0.100","payload":"0"},"result":{"media":{"callid":"0003e30c-c50c0134-763a6f53-359481cc at 192.168.0.100","mediaid":"1","tag":"0003e30cc50c5d43416d30b4-7b2f969c","payload":0,"rxbytes":108008800,"rxpackets":677772,"txbytes":108262240,"txpackets":676639,"local":{"ip":"192.168.1.233","rtp":50486,"rtcp":50487},"remote":{"ip":"192.168.5.1","rtp":5038,"rtcp":5039}},"media":{"callid":"0003e30c-c50c0134-763a6f53-359481cc at 192.168.0.100","mediaid":"1","tag":"2e268de3","payload":0,"rxbytes":108262240,"rxpackets":676639,"txbytes":108008800,"txpackets":677772,"local":{"ip":"192.168.1.233","rtp":34732,"rtcp":34733},"remote":{"ip":"192.168.5.10","rtp":20912,"rtcp":null}}}}
+
+
+Logging
+* Events logging (start, stop, interim update, timeout, codec/parser error) to syslog (with full RTCP printing)
+
+
+
+
+
+
+
+
+
 
UserCommentTimeVersion
cervajs2013-04-09 12:422
cervajs2013-04-09 11:311

MediaProxy

$
0
0
Diff from version 8 to 9 (current)

@@ -1,4 +1,9 @@

 !Media Proxy
+
+((MediaProxy Features))
+
+((MediaProxy Comparison))
+
 
 [http://www.ag-projects.com/MediaProxy.html]
 
UserCommentTimeVersion
cervajs2013-04-09 12:489

VOIP sites

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Diff from version 1834 to 1835 (current)

@@ -194,8 +194,7 @@

 *[http://freevoipcalls.ilikeit.asia|VoIP Blog]: A weblog dedicated to VoIP calling and latest news about VoIP services.
 *[http://resourcelibrary.voipinnovations.com/| http://www.VoIPInnovations.com] VoIP resource library. A place to learn VoIP Industry buzz words
 *[http://voipnews.pl|VoIP News] - Polish website with information about VoIP and telecommunication. Rating and comparison VoIP operators.
-*[http://www.voip-forums.info | http://www.voip-forums.info] VoIP and Video Conference Forum, update technology about VoIP, daily free technology report for member, support free forum for voip manufactures
-*[http://www.vnvoip.info | http://www.vnvoip.info] VoIP and Video Conference Forum, update technology about VoIP, daily free technology report for member, support free forum for voip manufactures, support many language
+*[http://www.voip.com.vn | http://www.voip.com.vn] VoIP and Video Conference Forum, update technology about VoIP, daily free technology report for member, support free forum for voip manufactures, support many language
 *[http://www.123world.com/VOIP/|VoIP|id=VoIP] Listings of VoIP sites, VoIP services, & VoIP blogs
 *[http://www.voip88.com] support Chinese asterisk/elastix community.
 *[http://www.voip-catalog.com] - VoIP Catalog - Online database of VoIP providers and companies
UserCommentTimeVersion
fone4vn2013-04-09 14:031835

Analog Telephone Adapters

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0
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Diff from version 458 to 459 (current)

@@ -73,7 +73,6 @@

 * ((Oki Network Technologies)) - FXS gateways
 * ((Ooma Telo))
 * ((Patton)) - ((M-ATA))
-* [http://www.pikatechnologies.com/english/view.asp?x=1210|PIKA Technologies Inc.] Mosport S Series connects up to 24 analog phones and/or faxes to a broadband modem or LAN
 * ((PLANET)): - [http://www.planet.com.tw/en/product/product_category.php?mt=menu_product_372&id=372|IP Telephony] - Deliver Your Voice in Real Time
 * [http://www.plextel.net|Plextel] - 1,2,4 ports ATA, SIP/H.323/IAX support
 * Pluscom SIP VoIP ATA - 1 FXS and 1 FXO. DOES NOT gateway from VoIP to PSTN or vice-versa (hence listing here). Avoid.
UserCommentTimeVersion
mennam2013-04-10 15:40459

VoIP Gateways

$
0
0
Diff from version 671 to 673 (current)

@@ -64,10 +64,7 @@

 * [http://www.obihai.com/product-primer.html|Obihai] - OBi110 : 1 FXS, 1 FXO, 2 SIP, OBiTALK P2P, AA
 * ((Oki Network Technologies)) - FXS & FXO
 * [http://www.ovislink.com.tw/products.htm#voip|Ovislink] - FXS & FXO, 2 to 8 port models
-
 * ((Patton Electronics)) - [http://www.patton.com/products/pe_products.asp?category=5&#catshow|SmartNode] - FXS & FXO VoIP gateways, routers and IADs, 2 to 32 ports. And you always get [http://www.patton.com/support/index.asp?h1=Free#Free | free support]!
-
-* [http://www.pikatechnologies.com/english/view.asp?x=1209|PIKA Technologies Inc.] Mosport M Series gateways featuring R2 CAS E1, ISDN E1 PRI, T1 PRI, BRI, or FXS/FXO interfaces
 * [http://www.pikatechnologies.com/english/view.asp?x=607|PIKA Technologies Inc.] - PIKA Gateway Boards for Asterisk
 * ((PLANET)): - [http://www.planet.com.tw/en/product/product_category.php?mt=menu_product_372&id=372|IP Telephony] - Deliver Your Voice in Real Time
 * Pluscom SIP VoIP ATA - 1 FXS & 1 FXO. Does NOT do gateway routing (i.e. from VoIP to PSTN or vice-versa). AVOID!

@@ -126,11 +123,7 @@

 * ((Multitech)) - BRI H.323 & SIP
 * [http://www.novatec.de/cms/en/products/transnovafamily/tip/index.html|Novatec] - VoIP-ISDN (BRI and PRI) -GSM-Media-Gateways
 * Parlay VoXip SIP Gateway - Including Japanese INS64 and LCR [http://www.voxtream.com]
-
 * ((Patton Electronics)) - [http://www.patton.com/products/voip-comparison.asp | Smartnode] BRI VoIP Gateways, routers and IADs, 1 to 36 calls. And you always get [http://www.patton.com/support/index.asp?h1=Free#Free | free support]!
-
-* [http://www.pikatechnologies.com/english/view.asp?x=1211|PIKA Technologies Inc.] - Mosport B Series digital ISDN BRI interface gateways
-* [http://www.pikatechnologies.com/english/view.asp?x=1209|PIKA Technologies Inc.] Mosport M Series gateways featuring R2 CAS E1, ISDN E1 PRI, T1 PRI, BRI, or FXS/FXO interfaces
 * [http://www.ratotec.com|Ratotec Hybrid Switch ] - ISDN VoIP Least Cost Gateway 4xS0 / 8xS0
 * [http://www.tas.de/index.php?id=17&L=1|TAS] - YPSILON survivable IN- and OUT Media Gateway for SIP and European ISDN; 2FXO + 2FXS
 * ((Telebau Phones)): 1-6 x S0 (ISDN BRI) gateway to LAN (H.323)

@@ -171,10 +164,7 @@

 * ((Nuera)) up to 2000+ phone lines
 * ((Oki Network Technologies)) - PRI, T1, & E1
 * Parlay VoXip SIP Gateway - Including Japanese INS1500 and LCR [http://www.voxtream.com]
-
 * ((Patton Electronics)) - [http://www.patton.com/products/voip-comparison.asp | SmartNode] - PRI/T1/E1 VoIP gateways, routers and IADs (15 to 120 calls). And you always get [http://www.patton.com/support/index.asp?h1=Free#Free | free support]!
-
-* [http://www.pikatechnologies.com/english/view.asp?x=1209|PIKA Technologies Inc.] Mosport M Series gateways featuring R2 CAS E1, ISDN E1 PRI, T1 PRI, BRI, or FXS/FXO interfaces.
 * ((quintum)) 1 to 32 E1 or T1 port models
 *[http://realtonetech.com| Realtone technoloy] 1,2,4 Digital ISDN PRI truning gateway for calling card, call termination, call center. PBX to VoIP connection; support SIP and T.38, iLBC codec, RADIUS client
 * Ratotec Hybrid Switch [http://www.ratotec.com] ISDN VoIP Least Cost Gateway HS-30
UserCommentTimeVersion
mennam2013-04-10 15:42673
mennam2013-04-10 15:39672

Atlanta VoIP

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Diff from version 5 to 6 (current)

@@ -1,5 +1,6 @@

 This is a list of VoIP providers in Atlanta. __Atlanta VoIP__ companies typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Atlanta to the list below.
 
+* [http://www.avoxi.com|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 like [http://www.avoxi.com/cloud-based-contact-center/|Cloud Based Contact Center], [http://www.avoxi.com/freephone-numbers/|Freephone Numbers], [http://www.avoxi.com/hosted-pbx-providers/|Hosted PBX Providers], [http://www.avoxi.com/international-toll-free-number-service-itfs/|International Toll Free Number] and [http://www.avoxi.com/voip-call-center/|VoIP Call Center]
 *[http://www.sipsrus.com| SIPSRUS] SIPSRUS provides cloud computing based business phone and fax systems designed for today's mobile and distributed business world. SIPSRUS provides services in USA, Canada, UK, Australia, Japan, Spain, India, UAE and many more!
 * ((Jive Communications)) - Jive offers Atlanta VoIP services business.
 * [http://www.Lantelligence.com/|Lantelligence] is an International provider of Business Communications solutions that include IP Phone Systems, Call Centers, Video and Web Conferencing services for multi-location corporations.
UserCommentTimeVersion
avoxi2013-04-10 19:556
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