VOIP-info.org Wiki Changes

voip-info.org

stygmah — Tue, 09 Feb 2010 17:51:11 +0100

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 & 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.

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NEWS

2010-02-09 - Squire Technologies Launch SMSC and SMS Gateway at MWC 2010
2010-02-09 - OwnPages releases real time patch for AsteriskGUI 2.0 Mysql or postgres - better manageability, easier integration.
2010-02-09 - Quobis and Asipto work together to market and support SIP solutions for telecom operators and governments in Latam and Iberian markets
2010-02-09 - The SIP School™ at UCEXPO London in March SIP Trunking Seminar - "Getting it right the 1st time"
2010-02-09 - Rho releases version 3.3 of AsteriskC2D and AsteriskC2DPro (iPhone click to call for Asterisk), new version has native Thirdlane support
2010-02-08 - US Patent Office Agrees to review C2/Acceris controversial VoIP patent
2010-02-08 - How to configure a Patton 4114 FXO Gateway with 3CX video tutorial
2010-02-06 - Anveo unveils Voice 2.0 communication suite with powerful Visual Call Flow technology. Visually create Voice IVR applications without any special skills.
2010-02-05 - Unified Recording release Open Unified Recording Chagall'. The Gigaset IP phone series is targeted towards home installations. However, the handsets can be used in larger, HiCom based enviroments as well. There are three categories of handsets, C-series (value), S-series (full featured), and SL-series (full featured and smaller sized).

Gigaset C450IP / C460IP

The Gigaset C450IP is a DECT dual mode SIP/PSTN phone. It has been designed for the consumer market but it easy integration in Asterisk makes it very applicable to professional installations. As it contains LGPL'd software (libOSIP) one can rebuild the firmware with a new libOSIP.

Gigaset C470IP / C475IP

The Gigaset C470IP supersedes the C450IP, featuring an improved handset. The C475IP also features an integrated answering machine, and supports 6 SIP account registrations

Gigaset S450IP

The Gigaset S450IP is similar to the C450IP, but uses a different handset and an almost identical yet more powerful (RAM/ROM size, CPU speed) hardware base.

Gigaset S675IP & S685IP

The Gigaset S675IP and S685IP seem to be a successors to the S450IP, it uses an improved handset and adds an integrated answering machine. The 675 was never officially imported into the UK, instead the 685 is the same except for adding Bluetooth to the handset for hands-free and receiving contacts in vCard format. Compared to the S450IP they use an improved handset and add an integrated answering machine.

Gigaset SL75 WLAN WLAN/SIP

The Siemens Gigaset SL75 WLAN is not based on the DECT series. It is a Wifi telephone with a camera. You can also read e-mails with it or use it as an IMS. The device seems to be running linux so it is probable that it will do much more than that.

OptiPoint Series

According to the datasheet and user manuals the phones support standard SIP as well as features of the HiPath8000 system, but also BroadSoft and Sylantro can be selected ("server type"). For all systems various business features are supported (keyset, call park, pickup, 3PCC...). The phones do not support STUN, but features like DNS SRV and survivability, QoS and extensive trace capabilities.
Actual application SW version is V6.0, unfortunately there is no SW in the public download area...
A WiFi with SIP protocol is also available: optiPoint WL 2 professional S.

Provisioning and maintenance can be done by either the (free-of-charge) tool "Deployment Service" (this can be downloaded from their web site www.siemens.com/hipath) or by (XML) configuration files or via local menue / web admin pages.

Refer to http://www. ...

VOIP Service Providers B2B

hansjuergen — Tue, 09 Feb 2010 15:45:34 +0100

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

VOIP Service Providers Business - Small office plans & PBX systems go here
VOIP Service Providers Residential - Residential services go here
Local Number Portability

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

Asterisk/SIP Service Providers
- North America
- Asia
- Australia / New Zealand
- Caribbean
  - Dominican Republic
- Europe
  - Austria
  - Belgium
  - Bulgaria
  - Cyprus
  - Czech Republic
  - Denmark
  - Estonia
  - Finland
  - France
  - Germany
  - Hungary
  - Italy
  - Ireland
  - Latvia
  - Lithuania
  - Luxemburg
  - Netherlands
  - Norway
  - Poland
  - Sangoma
    
    VoilensP — Tue, 09 Feb 2010 15:39:17 +0100
    
    http://www.sangoma.com/products_and_solutions/
    Sangoma has offered premium telephony solutions since 1984. This includes communications cards for PCs and other devices and now includes a set of software building blocks that talk SIP.
    
    Asterisk, Freeswitch, CallWeaver, Yate, 3CX, and pbxnsip compatible products.
    
    New Sangoma A-Series hardware now has a lifetime warranty
    
    SIP to TDM VoIP Gateway Cards
    - NetBorder Express
    NetBorder Suite for Call Centers
    - NetBorder Call Analyzer: Confidence-based Call Progress Analysis Engine (CPA)
    ISDN BRI Cards
    - A500 up to 24 ports of BRI PCIe or PCIx card, hardware echo cancellation optional
    - B700 FlexBRI hybrid card: 4 BRI ports and 2 FXS/FXO analog ports with Octasic HWEC
    ISDN PRI (T1/E1) Voice and Data Networking Cards
    - A101 one port T1/E1 PCIx card

Digium®, Inc., the Asterisk® company, is the original creator and primary developer of Asterisk®, the industry's first open source telephony platform. Digium provides hardware and software products, including AsteriskNOW™, the complete open source software appliance; Asterisk Business Edition™, the professional-grade version of Asterisk; and the Asterisk Appliance™, hardware-based telephony solution, to enterprises and telecommunications providers worldwide. Digium also offers a full range of professional services, including consulting, technical support, and custom software development.

Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.

History

Mark Spencer founded Linux Support Services in 1999 while still a Computer Engineering student at Auburn University. When faced with the high cost of buying a PBX, Mark simply used his Linux PC and knowledge of C code to write his own! This was the beginning of the world-wide phenomenon known as Asterisk, the open source PBX, and caused Mark to shift his business focus from Linux support to supporting Asterisk and opening up the telecom market. Linux Support Services is now known as Digium, and is bringing open source to the telecom market while gaining a foothold in the telecom industry.

Digium is based in Huntsville, Alabama.

Philosophy

Digium is a young and fast-moving company with an energetic and hip culture that reflects the philosophy of Asterisk and the open source revolution. Mark strongly believes that every technology he creates should be given back to the community. This is why Asterisk is fully open source. Today that model has allowed Asterisk to remain available free of charge, while it has become as robust as the leading and most-expensive PBXs.

Digium was founded on the notion that the customer should have control over the technology that goes into his telecommunications systems, a rare notion in the proprietary world of Telecom. Since the inception of Asterisk, the idea of open source communications technology has been revitalizing an industry which was at one time crippled by the dominance of monolithic dinosaurs.

Digium employees find an added joy in their work knowing that they are pioneering this revolution. The Digium offices are filled with innovative problem-solvers in every department, not just in engineering, because fresh thought has been key to Digium's success from the very beginning.

Website: http://www.digium.com
Store: Digium Store
Sales: sales@digium.com
Support: support@digium.com
Telephone Local: (256) 428-6000
Toll Free: (877) 344-4861 (877-DIGIUM1)
IAXTel: (700) 428-6000

Job Openings:

Work@Digium

Products:

Complete List

Hardware

Joseph Arsenault - Last Major clean up of list on June 17th,2009

Here's a list of the Cheapest SIP Outbound calls, DID's Per country and ATA Adapters

Page Contents

Free PSTN DIDs & Minutes - This is ONLY for free (as in no catch) PSTN DIDs & Minutes.
- Shared DIDs (You call a central number, and put in your Extension, or Enum):
- Free Minutes to Various Countries
PSTN DIDs (Ranking: By Country, Provider Name)
Minutes (Tiered rates to US48)
Minutes (Ranking: By Country, Termination Cost)
Unlimited Plans (Ranking: By Country, Provider Name)
ATAs - NO LINKING TO HOME PAGE! ONLY TO PRODUCT PAGE
Phones- NO LINKING TO HOME PAGE! ONLY TO PRODUCT PAGE

Rules
DO NOT remove your competitors unless you are cheaper than they are (AFTER shipping)
DO NOT post your store's front page, only linking directly to the product mentioned will be allowed (You WILL be deleted for this one)
DO post your prices (listings without prices will be removed without even checking if you're the cheapest)
DO link to your store, but be aware that if you're not the cheapest you might be removed.
DO sort listings with cheapest listing on top (if more than one is provided)
And please add REFURBISHED products as ANOTHER product (see Sipura SPA-2000 example, below).

When comparing prices on hardware, it's wise to check the final price, including shipping and taxes.

Free PSTN DIDs & Minutes - This is ONLY for free (as in no catch) PSTN DIDs & Minutes.

Romania GeoTel Low cost calls and free dial in numbers in Romania.
UK - Localphone Localphone provides free geographic number of most of UK cities just by registering for SIP account. Their service is reliable and call charges are very low.
USA - www.fonosip.com FonoSIP USA DIDs - Free VoIP Account plus DID area code 206 253 360 or 425
USA - www.ipcomms.net FREE USA DIDs - 1 Number + 2 Lines = FREE SIP Delivery
USA - IPKall Free DID's in USA 206, 253, 360, 425 (Seattle/Tacoma WA area)
USA -

Page Contents

ConfBridge
- Synopsis
- Description
Asterisk cmd ConfBridge
- Muteing
- Technical Details (for developers)

ConfBridge

Synopsis

ConfBridge conferencing bridge

Description

ConfBridge([confno][,[options][,pin]]): Enters the user into a specified ConfBridge conference

Asterisk cmd ConfBridge

ConfBridge is an application for Asterisk starting with the 1.6.2.* series. ConfBridge is very similar in features to MeetMe, but unlike MeetMe, ConfBridge does not perform audio mixing using DAHDI. Instead, audio mixing is performed within the internals of Asterisk.

To get an up2date description of ConfBridge for your used Asterisk version execute "core show application ConfBridge" on the Asterisk CLI.

The option string may contain zero or more of the following characters:

'a' — Set admin mode
'A' — Set marked mode
'c' — Announce user(s) count on joining a conference.
'm' — Set initially muted.
'M' — Enable music on hold when the conference has a single caller. Optionally, specify a musiconhold class to use. If one is not provided, it will use the channel's currently set music class, or 'default'
'1' — Do not play message when first person enters
's' — Present menu (user or admin) when '*' is received (send to menu)
'w' — Wait until the marked user enters the conference
'q' — Quiet mode (don't play enter/leave sounds).

Muteing

When a participant is "muted" this means that the participant's audio is ignored. Nevertheless the muted participant still receives the mixed audio stream.

Technical Details (for developers)

ConfBridge() uses Asterisk's bridging framework which was introduced in Asterisk 1.6.2. There is a common bridging framework in main/bridging.c. The bridging implemenations itself are modules located in bridges/bridge_*.c. There are several implementations with different kind of features. When an application requests a new bridge (e.g. ConfBridge), it has to specify the needed features and the bridging framework will choose the best fitting bridging implementation.

VOIP Service Providers Residential

hansjuergen — Tue, 09 Feb 2010 14:34:04 +0100

Service Providers Residential

Africa

Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
IPtransit Residential and SME Sip, H323, SIP URI, Inbound, LoCall Plans, PC and Mobile Phone clients, well suited to VSAT, SCPC and contended bandwidth clients, 723.1, &29, 711 and true T38, Calling cards and GSM Gateways, BYOD welcome, test accounts.
Corelynx Inc Hosted Enterprise Telephony for SME and Residential, Full Suite of Call Center Solution - Hosted and Onsite Model, Offers DID and 8XX numbers to more than 50 countries of the world. Capable of offering VOIP based IP PBX Solution on MPLS and VPN for countries where VOIP ports are blocked, IPLC and Colocation Services, A-Z wholesale and retail termination. Also offers any form of customization work on Asterisk or SER platforms.
INDIGICOM VoIP Solutions Provider. Business services, end-users, satellite VoIP.
MyKanKan VoIP Service Provider. Business solutions for call-centers, hotels, callshop .... Voip and Asterisk consulting. Features include : Voicemail, Caller ID w/Name, Call Waiting, Call Forwarding, Caller ID Block, Free In-Network Calling, Web Based Call Logs. Optional Services include: SoftPhone Access, Virtual Phone Number.
http://www.powerbillbox.com African VoIP carrier and billing provider. African PSTN calling and peering, national & international rates in H323, SIP, IAX.
Takalam Takalam offers VoIP solution starting from $10 to small and medium business in all African and Arab countries like Saudi Arabia, UAE, Jordan, Egypt, Bahrain, Kuwait, Iraq, Syria, Palestine, Kenya, Nigeria, Rwanda, etc...
TuT Telecom - ABSOLUTELY FREE Peer-to-peer SIP calls vis softphones or SIP hardware. Local DIDs also available to over 7000 cities. Asterisk Supported. PC2Phone services and voip services. WiFi & GSM Voip Phones and services available.

South Africa

Asterisk Directory South Africa Comprehensive Listing of Asterisk Companies and VoIP providers in South Africa
Logic Ring Very competitive International VoIP Reseller programs for agents and private label resellers.
FutureFone South African VOIP provider. SIP based account. ...

DID Service Providers

hansjuergen — Tue, 09 Feb 2010 14:01:14 +0100

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

Free Service Providers Only (Free DIDs)

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.
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.
Actio.pl Polish business class VoIP service provider for enterprise, contact center and call center clients. Free polish DIDs.
FonoSIP Free sip accounts and US DIDs, automatic activation.
Netelip Free spanish DIDs. No Setup Fees. Porting allowed.
TandemTransit.com Unlimited DIDs within LATA 132 NYC. IP ports and Transit also available. No Setup Fees. Porting allowed.
DIDLogic.com New DID trading platform. SIP/IAX/H323. No fees. FREE Gtalk, Skype forwarding! Full time https.
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.
3Stars Net FREE Belgian DIDs in all Belgian areas. Also provides premium and 0800 numbers in Belgium. (The service is not free anymore)
http://www.audmaster.comFREE call forwarding of Virtual numbers to your PBX, Calling Card gateway. Free 2USD credit to test service on registration
aql, Free Signup with PSTN 0870 UK number
IVR from Wikipedia:
- "In telephony, interactive voice response, or IVR, is a phone technology that allows a computer to detect voice and touch tones using a normal phone call. The IVR system can respond with pre-recorded or dynamically generated audio to further direct callers on how to proceed. IVR systems can be used to control almost any function where the interface can be broken down into a series of simple menu choices. Once constructed IVR systems generally scale well to handle large call volumes."
IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component
IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone keypresses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration
IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.
- Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
- A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.
One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting
IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchphone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Graphical Design Tool for IVR Applications
Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

See Also (Vendor Information)
IVR Information
- CCXML standard markup language for IVR / call control applications
- Examples of

VOIP-info.org Wiki Changes

voip-info.org

Welcome to the VOIP Wiki - a reference guide to all things VOIP

NEWS

Gigaset C450IP / C460IP

Gigaset C470IP / C475IP

Gigaset S450IP

Gigaset S675IP & S685IP

Gigaset SL75 WLAN WLAN/SIP

OptiPoint Series

VOIP Service Providers B2B

Page Contents

Sangoma

History

Philosophy

Job Openings:

Products:

Hardware

Here's a list of the Cheapest SIP Outbound calls, DID's Per country and ATA Adapters

Page Contents

Free PSTN DIDs & Minutes - This is ONLY for free (as in no catch) PSTN DIDs & Minutes.

Page Contents

ConfBridge

Synopsis

Description

Asterisk cmd ConfBridge

Muteing

Technical Details (for developers)

VOIP Service Providers Residential

Service Providers Residential

Africa

South Africa

DID Service Providers

Cheapest DID Providers

Free Service Providers Only (Free DIDs)

Computer Telephony Component

IVR and ACD Integration

IVR and Voice Broadcasting

Graphical Design Tool for IVR Applications

See Also (Vendor Information)

IVR Information