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  • 12/15/15--03:12: VoIP Termination
  • Please add information to this page about VoIP call termination.

    What is VoIP Termination?

    VoIP call termination is used to refer to the procedures that are used for routing telephone calls from one provider to the next provider until the call has been routed to the last telephone company and has been received by the recipient. Voice termination is another term that is used for call termination. The telephone companies are also referred to as providers or carriers.

    Called Party

    The called party is the person who has received the telephone call. The end point of the route may be on the Internet or may be at a point that was reached by routing the call through the public switched telephone network. The procedures for routing the call stop when the call has been received by the recipient. The process may seem simple to individuals who do not experience problems with making telephone calls but is not so simple to individuals who make calls that will not connect such as calls to a different country.

    Calling Party

    The calling party is the person who has initiated the call and who wants to be connected to the called party. There may be problems with the telephone calls that begin on the Internet and end at a cellular phone. The sound of the voices may seem to be amplified with echoes and have a scratchy quality. The calls that are initiated with Google Voice may never connect or could be delayed.

    VoIP

    Voice over Internet protocol (VoIP) is a term that is used to describe a call that was initiated on the Internet such as Skype calls or calls made with Google Voice. The calls that were initiated on the Internet usually end at a point that is not on the Internet. Most of the recipients of telephone calls receive the calls with a landline phone or with a cellular phone. The route may begin on the Internet but will end at a point that was reached by routing the call from the public switched telephone network (PSTN), which is the common description for call termination.

    Internet Networks

    A tier-one operator is licensed and registered to operate an Internet protocol (IP) network for Internet telephony services. Tier-one operators can handle call origination and call termination. A tier-two operator can lease services from a tier-one operator. The tier-three operators can lease services from either a tier-one or two-two operator. There are also resellers of VoIP services and wholesalers in the market. The quality of the services is not very high because of the inconsistencies in the market such as fluctuations with demand, fraud and problems with doing business on an international level.

    Call Origination

    Call origination is used to refer to telephone calls that originate from the public switched telephone network and end the route on the Internet. Call termination is considered to be the opposite of call origination because the direction of the paths are reversed. The terms are associated with the starting point of the calls, path of the route of the calls and termination point of the calls. The operators of IP telephony services can handle calls that originate or terminate on the Internet.

    Fees

    The fees for services are subject to the regulations of several countries because the routes of the calls will cross over more than one country. Those countries can use legislation to control the fees for the services. Termination rates are usually very high for the countries in the Middle East and Asia. The rates are intentional high because there are more incoming calls than outgoing calls from the countries, which is caused by the diaspora effect of a migrating population.

    VoIP Termination Providers

    Please list VoIP Termination providers here in alphabetical order.

    10gea 10gea's wholesale SIP termination provides exceptional quality routes and high volume switching capacity for all types of Wholesale end users. Very competitive rates for both dialer and conversational high volume traffic on tier one routes. ...

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  • 12/15/15--04:15: Emergency Calls
  • Let's build a list of all emergency numbers for the world!


    Not everybody uses 911. You'll realise when you need it most
    911 is used e.g. in USA, Canada, Brazil, Salvador, Costa Rica, Cameroon, Aruba
    999 is used e.g. in Malaysia, Singapore, Qatar, Malaysia, United Arab Emirates, UK
    So please add your country to this page NOW

    Rule of thumb: 911 America, 112 Europe

    Details see below

    Australia

    000 All emergency services (Police, Ambulance and Firebrigade)

    Austria

    112 European emergency number. Dispatch to all kind of emergencies (see below)

    122 Firebrigade
    133 Police
    144 Ambulance
    140 Alpine rescue
    141 Medical emergencies


    Belgium

    112 European emergency number
    100 Firebrigade, Ambulance
    101 Police

    Brazil

    190 Police
    192 Ambulance
    1923 Firebrigade

    China

    110 Police
    119 Firebrigade
    120 Ambulance

    Denmark

    112 European emergency number
    114 Local Police

    Europe

    112 The Europe-wide emergency number for police, fire and medical emergencies. Valid in all member states of the European Union as well as Iceland, Liechtenstein, Norway and Switzerland.

    France

    15 Ambulance
    17 Police
    18 Firebrigade

    Germany

    112 European emergency number
    110 Police

    Greece

    100 Police
    166 Ambulance
    199 Firebrigade

    Japan

    110 Police
    119 Firebrigade, Ambulance

    New Zealand

    111 All emergency services (Police, Ambulance and Firebrigade)

    Singapore

    999 Police
    995 Firebrigade, Ambulance

    Sweden

    112 European emergency number

    Switzerland

    112 European emergency number. Dispatch to all kind of emergencies (see below)

    in former times you called theses specific numbers
    117 Police
    118 Firebrigade
    140 Emergency road service
    144 Ambulance
    145 Poisoning
    1414 Alpine rescue (by helicopter)

    Thailand

    191 Police
    199 Firebrigade
    1699 Emergency number for tourists

    UK

    112 European emergency number
    999 Police, Firebrigade, Ambulance, Coastguard

    Ofcom amended General Condition 4 on 8 September 2008. From this date, all VoIP services that allow users to make calls to traditional fixed or mobile phones (type 2 VoIP services), or to make calls to and receive calls from traditional fixed or mobile phones (type 4 services) must provide 999 / 112 access at no charge. "Click to call" services are excluded from this obligation.

    Furthermore, the Network Operator must also provide Caller Location Information for calls to the emergency call numbers, to the extent that is technically feasible. These requirements already apply to fixed line and mobile communications providers. In its policy statement of 5 December 2007 (http://www.ofcom.org.uk/consult/condocs/voip/voipstatement/voipstatement.pdf) Ofcom stated that, in relation to VoIP calls, ‘technically feasible’ should be taken to mean that location information need only be provided where the VoIP service is being used at a predominantly fixed location.



    See Also



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  • 12/17/15--05:47: Sip Trunking Providers
  • This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

    Country specific pages:


    What Is SIP Trunking?

    Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

    One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

    The Benefits of Using SIP Trunking Services

    Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
    • It offers very low cost calling.
    • It's much easier to scale than other options, making it very future proof.
    • SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
    • Network outages are much less impactful, as incoming calls can easily be routed to other locations.
    • It's ideal for any sized business with at least 25 physical phones.
    • It's a fantastic choice for any business that has an international location.
    • It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

    How SIP Trunking Can Take Your Business To The Next Level

    It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

    Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

    All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level. ...

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  • 12/17/15--05:53: new VOIP services
  • This is a list of new VoIP services.

    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.

    December 2015


    February 2015


    July 2013


    June 2013


    May 2013


    April 2013


    March 2013

    • 2013-03-01 -

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  • 12/17/15--10:05: Asterisk Operator Panel
  • We've been working on a new open source asterisk panel:

    • Can connect multiple asterisk servers
    • Flexible support for multi tennant environments
    • Minimal configuration
    • Auto detection of SIP devices
    • Web based, no flash required.
    • Login is done by calling a special number, so no additional password and user management is neccesary.
    • Tested with configurations generated by:
      • Asterisk_GUI
      • Thirdlane.
      • FreePBX


    Project page http://open.syn-3.nl/syn3/trac/default/wiki/projects/synapse#Automaticasteriskoperatorpanel

    Source code: https://github.com/psy0rz/Synapse

    We just started beta testing with our customers, so documentation is lacking quite a bit.

    If you need any help, mail me at edwin@datux.nl



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    Virtual private servers (VPS) or Virtual dedicated servers are a form of virtualization that splits a single physical server into multiple virtual servers. The practice of partitioning a single server so that it appeared as multiple servers has long been common practice in mainframe computers, but has seen a resurgence lately with the development of software such as VMware, User-mode Linux, FreeVPS, Virtuozzo and Xen. (From Wikipedia)

    At the outset, possible issues arise out of inability or difficulty in sharing timing source (required for MeetMe). linode has some suggestions


    Virtual Private Asterisk Service Providers

    • AJTEL VIRTUAL Es el servidor de hosting FreePBX mas completo del mundo ahora a tu alcance. Este servicio de alojamiento es exactamente como tener su propio Asterisk privado y servidor FreePBX con un minimo costo inicial y SIN NINGUNA dificultad técnica. También obtiene todos los beneficios de tener un servidor en un entorno de centro de datos de clase mundial. Además tenemos versiones para Empresas muy pequeñas y servidorees dedicados para grandes Empresas. y ademas pagando... en tu Propia moneda en Mexico y/o en USA.
    • PowerPBX Private Asterisk & FreePBX server hosting. This hosting service is exactly like having your own private Asterisk & FreePBX server without the upfront cost and technical difficulty. You also get all the benefits of having a server in a world class datacenter environment. We offer all the popular distributions as well as our own highly optimized lean mean telephony machine.
    • FreePBXhosting.com FreePBXhosting.com is the ONLY FreePBX hosting provider approved by SchmoozeCom & FreePBX.org. Our Virtual Private Server options are the perfect for beginners and experienced FreePBX users. Simply pricing structure along with fully automated setup, backup and restore options. Upgrade as you grow. Our Premium Dedicated Servers allow you run on your own dedicated hardware without the hassle of owning the hardware. All of our dedicated servers are fully configurable during the ordering process. We also offer High Availability Dedicated Server configurations allow you run a High Availability environment without the hassle of owning the hardware and configuring the networking.
    • Iguanahosting.com/elastix Asterisk - Elastix Hosted Dedicated Servers, Neutral Bandwidth on our DC @Atlanta ,GA 100% Uptime, Load Balancing Available.
    • LYLIX provides unmanaged VPS hosting with complete Asterisk support on any of eight Linux distributions in addition to Trixbox, Elastix, AsteriskNow, and PBX-in-a-Flash. Infrastructure backed by a High-availability (HA) network over three regional NOCs (USA). Meetme, MOH, g729, Cepstral TTS, and Lumenvox capable.
    • Phonewire managed and unmanaged VPS hosting with Asterisk, Trixbox, and PBX-in-a-Flash install images available.
    • RentPBX VPS hosting with full root access and dedicated ip with Asterisk, AsteriskNow, Elastix, Trixbox, and PBX-in-a-Flash, A2Billing, Sipxecs, Vicidial and Blue.Box install images available. Atlanta, Ashburn, Dallas, Chicago, Los Angeles, Miami, Mountain View, New Jersey, Seattle and Tampa Bay nodes are available. We have available nodes in Germany, Toronto & Montreal Canada and United Kingdom. We specialized in IP-PBX VPS only. We only offer PBX hosting. $19.99/ month with coupon NEW52011. ...

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  • 12/22/15--12:54: IP PBX
  • IP PBX is a phone system that utilizes IP communications. Traditionally IP PBX's are located on site where they can also interface to traditional telco services such as analogue phone lines. The business end users connect via IP to the IP PBX for voice service.


    What is an IP PBX?

    An IP PBX can be referred to as a lot of things: a business phone system, a unified communication system, or simply as a "PBX. ...

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  • 12/22/15--21:17: Asterisk
  • Image

    Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and OS X and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

    Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H.323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny. Check the Features section for a more complete list.

    Asterisk needs no additional hardware for Voice-over-IP, although it does expect a non-standard driver that implements dummy hardware as a non-portable timing mechanism (for certain applications such as conferencing). A single (or multiple) VOIP provider(s) can be used for outgoing and/or incoming calls (outgoing and incoming calls can be handled through entirely different VOIP and/or telco providers)


    For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsor, Digium. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks . ...

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  • 12/22/15--21:35: FreeSwitch
  • What is it?

    FreeSWITCH™ is an open source communications platform. FreeSWITCH™ is a library which ships with a small executable that loads the library, launches the core, and performs the various tasks that are defined by the modules. In its base form FreeSWITCH™ is a soft-switch or PBX telephony application, not completely unlike Asterisk but capable of handling thousands of simultaneous calls.

    FreeSWITCH™ makes it possible to build a softphone, PBX system, soft-switch, or interface with other open source PBX systems such as OpenPBX.org, Bayonne, Yate or Asterisk. It can also be used to build a VoIP switching platform uniting various technologies such as SIP (using the Nokia Sofia library), H.323, SCCP, LDAP, Zeroconf, XMPP / Jingle, etc.

    As a library FreeSWITCH™ can be used by developers to enable switching in their custom applications. FreeSWITCH™ can be launched not only from a C application, but also via PHP, Perl, or a variety of other languages.

    FreeSWITCH™ is written in C, built from the ground up (not a fork of another code base). It is designed to take advantage of as many existing software libraries as possible. It has a modular, extensible architecture, with few and necessary functionality in core (libfreeswitch) with optional modules to do the rest.

    FreeSWITCH runs on Windows, Mac OS X, Linux, *BSD, ARM, and other Unix flavors.

    It is licensed under the MPL.

    Features

    • Runs on Win32/Mac/Unix
    • IVR API
    • 8kHz/16kHz/32kHz/48kHz audio
    • Soft conferencing
    • SIP B2UA/SRTP/TLS
    • SIP BLF/SLA/PBX features
    • Presence
    • Google Talk
    • IPv4/IPv6
    • ENUM/ISN
    • Async audio
    • Event/logger engine
    • Real time
    • zRTP (libzrtp)


    Supported protocols

    • SIP (Sofia-SIP)
    • Skinny Call Control Protocol (SCCP)
    • Google Talk (dingaling)
    • H.323 (OPAL) (beta)
    • Skype (Skypopen) Watch 4 part video of Giovanni Maruzzelli on Skypopen.

    VoIP Today News




      FreeSWITCH based Solution



      News Resources



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    • 12/24/15--04:12: VOIP Service Providers
    • For a list of VOIP to PSTN service providers, indexed by country, please see:


      VoIP and VoIP Service Providers

      What is VoIP?

      VoIP (which stands for "voice over internet protocol" and is commonly referred to simply as an internet phone) is a highly cost effective and reliable way for businesses or even homeowners to make calls across the world or even just across town. The majority of major cable companies that offer bundled internet, television, and phone services already utilize this newer technology, but there are tons of other independent companies that specialize in providing this service to their customers at reasonable rates and with tons of extra features.

      More Than Just Computers

      When people think of VoIP, they generally think of computers due to the popularity of the numerous free communication services like FaceTime and Skype, but this is truly just one aspect of what VoIP can truly offer. It is true that VoIP technology transmits voice communication that's been converted into digital data across a packet-switched network or the internet (what this means, in essence, is that a user making phone calls over high speed internet lines rather than phone lines). With that in mind, users are not confined to only using it on a computer. VoIP technology can connect through the internet using traditional telephone equipment just like a regular line. The phone itself is connected to the internet using an adaptor that's plugged straight into a home or business's internet network. Most major services offer a softphone option as well, which allows the user to use their computer directly as a telephone service. In addition to all that, VoIP providers will generally also offer mobile or tablet apps that allow their customers to make calls on the device (assuming it's connected to Wi-Fi at the time). ...

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    • 12/24/15--04:17: DID Service Providers
    • 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

      SMS enabled DID Providers

      • DIDWW - the source for wholesale International DIDs and Toll Free Virtual Numbers. We provide SMS enabled DIDs in Canada, Israel, Russia, UK, Ukraine and growing.
      • MultiTEL is providing retail and wholesale Worldwide DIDs - over 90 countries. Pick your own SMS enabled DIDs from over 40 countries. (US, UK, CA, Germany, etc). Free access to JSON and SOAP API provided along with programming samples. No contracts, pay per month, instant activation. SMS forwarded via e-mail , URL or as SMS/Text message. All calls are forwarded to SIP, PSTN or to our free Hosted PBX. Coverage and numbers always available in stock from more than 90 countries.

      DID Providers by country

      Algeria

      • CarryMyNumber.comAlgeria DID /Virtual Phone Numbers at _wholesale rate@$ 4/month with free PBX with the Free forwarding to SIP, Softswitches ,Trixbox ,Asterisk or VOIP. Phone Numbers from over 70 countries available. Free PBX . Unlimited Channel numbers for call centers /Calling Card Providers__. Largest FootPrint worldwide. No Per Minute charges.
      • BuyDDINumbers.com Provides Cheapest Algeria DID /Virtual Phone Numbers/DDI Numbers @_€ 6.95/month including free PBX. with the Free forwarding to Skype ,Gtalk , iTalkWorld, Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX __. Toll Free Number Available without Monthly commitments.
      • BuyDIDNumber We Provide Algeria Virtual Phone Numbers@ $ 7.99 / Month NO SETUP FEE , UNLIMITED CHANNELS available with the Free forwarding to Skype ,Gtalk , Trixbox ,Asterisk , voipbuster , iTalkWorld ,any Betamax Delmont Voip or any other ITSP . Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments.
      • iTalkWorld.com Currently only national Algerian numbers. Also 70 other countries available. Free forwarding to our PC & Mobile apps , Our PC and Mobile Apps also work in Countries where voip is Blocked . Further We have Lowest Call Forwarding Rates anywhere in world , starting as low as 1/2 cent.
      • divertmycalls.com Provides Cheapest Algeria DID /Virtual Phone Numbers @ $ 7.99 with the Free forwarding to Skype ,Gtalk , iTalkWorld, Trixbox ,Asterisk or VOIP. Phone Numbers all over the world are available. Free PBX . Toll Free Number Available without Monthly commitments.
      • Buyvirtualnumber. ...

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    • 12/24/15--04:25: VoIP Providers USA
    • This is a list of VoIP providers in the USA. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers USA to the list below.

      VoIP Providers in the USA

      • CHECKBOX, VoIP Carrier, Wholesale A-Z, CALL CENTER CARRIER - We provide VoIP connection specially for CALL CENTERS / MARKETING COMPANIES who needs High Stress / CPS routes, We have USA CC, Canada CC, Australia CC, China CC, UK CC and others, Direct Routes to many countries.
      • Vox Connect - Cloud based phone systems, cloud hosted PBX, SIP trunks Vox Connect is a cloud-based phone systems provider serving Chicago’s home, small and medium-sized businesses. We provide cloud hosted phone systems, hosted PBX ( also known as cloud based PBX ), domestic and international calling services, local/toll free numbers for USA and Canada.Our high-quality cost-effective telecom services are custom-designed to fit your business – and your budget.

      • Zaplee - Seamlessly integrate with Skype, GoogleTalk, VoIP, SIP, landline and mobile phones to make Zaplee fit into your existing framework easily. Forward and route calls to any line of your choosing, and keep track of new voicemails right in your email mailbox. Inbound calls to Skype and SIP extensions are FREE and Other are 2.3 cents per minute in the United States.

      • http://www.cryptovoip.in/ - We provide Software development and customization services in VoIP Domain. Our Product suite consists of WebRTC based Web Dialer, Web Click-to-Call, VoIP Mobile Dialers for Android, iPhone and Windows 8. PC VoIP Dialers for Windows and Mac. Tunneling Solution to make all VoIP Dialers work in VoIP and SIP Blocked areas like Dubai, Oman, France, etc. We customize our products and solution with your images and Brand. We also provide services like Custom Software Development and Maintenance and Offshore Development and OutSourcing. Please visit www.cryptovoip. ...

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    • 12/27/15--03:50: HOMER
    • homerlogo.png


      HOMER is a robust, carrier-grade, scalable SIP Capture system and Monitoring Application with HEP/HEP2, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box, ready to process & store insane amounts of signaling with instant search, end-to-end analysis and drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling.

      Homer can be downloaded & installed from SIP Capture GITHUB Repositories
      HOW-TO, FAQ and guides for OpenSIPS,Kamailio,
      Freeswitch and
      Asterisk


      Homer is composed of:


      homerflow1.png


      CAPTURE NODE(s):

      Collects/indexes mirrored SIP traffic to database
      Based on sipcapture module for industry standard Kamailio/OpenSER/OpenSIPS comes with powerful fine-tuning and filtering options, flawlessly handling millions of packets per hour

      CAPTURE AGENT(s):

      Captures/forwards packets to capture node
      Native IP Proto 4 support (ACME Packet, Hauwei) and port mirroring/monitoring
      Native capture support in Kamailio/SER & FreeSWITCH, any other supported via
      Homer's universal Capture Agent with HEP & Filtering for total flexibility

      WEBHOMER UI:

      User interface to Homer capture DB & API
      Search, filter and display SIP packets and SIP sessions details,
      Visualize end-to-end Call Flows, extract PCAPs, display traffic & statistics,
      Multi-user, Access Levels, internal/external authentication (LDAP,Radius)
      with a fast, drag & drop, customizable Ajax/jQuery dynamic interface

      • Thousands of packets/sec
      • Multiple database options
      • Powerful UI Search/Filter
      • Native Call-Flow & PCAP
      • Statistic Charts & Analytics
      • REST API & Widgets
      • Multi-User w/ LDAP, Radius
      • Drag & Drop UI
      • Firefox, Chrome & IE support
      • 100% Open Source
      • Daily updates & improvements


      WebHOMER 5 ships with a full-featured REST API, enabling applications and scripts developed in any language to interface with Homer SIP capture database and extract the same detail of information available using the user interface in real-time as JSON and XML for post-processing, storage or visualization


      howto.jpg



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    • 12/27/15--04:05: New Software Releases
    • This page is to inform on various VoIP related software releases.

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


      December 2015


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    • 12/29/15--05:22: Causes of Echo
    • Causes of echo


      In Brief:

      Its most likely your echo is caused by the far-end point. The loudness of the echo on VoIP calls is no worse than PSTN calls. The difference is that because of the inherent delay induced by VoIP, echo is much more noticeable.

      Remember, for echo to be noticeable it has to be both loud AND delayed. In the normal PSTN world echo is loud, but NOT delayed therefore you don't notice it. Its for this reason telcos almost NEVER have echo cancel for local calls (hardware echo cancellers are expensive so why put them where you don't need them?). This is why in many cases you will notice bad echo on local calls but not on long distance or calls to cell phones. Long distance and cell phones always have echo cancellation. Some local calls may also have it which is why echo is often intermittent.

      Solution: The only thing you can do is implement echo cancellation as close to the far end point as possible. Since you don't control the telco, most likely that is whatever you have connected to the PSTN (Asterisk box).

      This page http://www.aussievoip.com/wiki/index.php?page=EchoInfo has a very good explanation.


      The phenomenon known as echo - whereby one or both parties hear what they just said a few milliseconds later - is one that seems to affect VoIP users frequently. It can range from being mildly irritating, to being utterly unacceptable.

      It's important to fully understand the causes of this effect before you can take measures to eliminate it. This page attempts to explain clearly where echo comes from. Only once you've read and understood the explanation given here should you consider Asterisk echo cancellation.

      Typical telephone connection


      Leaving VoIP out of the equation for a moment, a typical peer-to-peer telephone conversation looks a lot like this:

      >)--<-+                              4-wire trunk                                     +->--(<
             |                       +---------------->--------------+                       |
             |   2-wire local loop   |                               |   2-wire local loop   |
          Hybrid <---------------> Hybrid                         Hybrid <---------------> Hybrid
             |                       |                               |                       |
             |                       +----------------<--------------+                       |
       o)-->-+                                                                               +-<--(o
      
       Caller                                                                                Callee
      


      This shows a (slightly simplified) connection, with two-wire segments between the two subscribers and their local exchanges, and a 4-wire (transmit and receive pairs) line between the exchanges, and within the telephones themselves. There are "hybrids" at each end to divide the 2-wire local line into the 4-wire trunk, and to connect the telephone microphone and earpiece to the 2-wire local lines.

      The 2-wire local loop is often known as a POTS (plain old telephone service) line.

      Sources of echo


      There are several points in the above system where echo can be introduced.

      From the caller's perspective, these are:

      • Within the caller's telephone; a certain amount of the signal from the microphone is fed straight back to the earpiece, called "sidetone." This is often done by design (see also echo and sidetone ), and in any case, is not a problem here - more on why later. A particular special-case of this is a poorly-configured analogue (eg TDM400P) card - for example, the default (FCC) is not suitable for the UK.
      • At the hybrid at the callee's end. An improperly balanced hybrid won't correctly filter out all of the transmitted signal, and will reflect some of it back down the trunk. Imbalance may be from poor design (common) or unpredictable impedance conditions on the POTS line (very common). ...

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    • 12/29/15--13:45: VOIP Consultants
    • 0 0

      Australia


      1Voice Services

      Tel:0403094345
      • Asterisk - Realtime, ODBC, Calling Card, Call Center, CRM Integration, Faxing, F2Email, SIP, IAX, FXO/FXS, DUNDi, Queues, Call Progress Detection, Virtualization
      • Cisco / Avaya / Asterisk / NEC / Genesys Integration
      • Custom dialplan / ODBC functions
      • Multi-Phone provisioning systems
      • ViciDial, GoautoDial, Puff Dial, Elastix Call Center, FreePBX
      • global DID / SIP termination
      • Hyper-V, VMWare, Azure, AWS
      • Free initial consultation
      mailto:astfiji@gmail.com

      3Play Networks

      Brisbane, QLD, Australia.
      Tel:1300 301 946
      • Asterisk implementation and support
      • Networking
      • Business TDM and VoIP Services
      • Business xDSL and Ethernet Services
      www.3playnetworks.com.au
      info@3playnetworks.com.au

      Amit Mehta

      Tel: +61451504435
      Asterisk PABX Implementation and Support
      A2billing and Asterisk Installation and Integration
      SIP trunk installation and integration
      Customised IVR Build and Integration
      Contact Center Specialist
      Vicidial and Go-Dial Integration
      OpenSer,Kazoo and Kamailio Specialist
      Experienced Technical staff available
      mailto:amit.magnate@gmail.com

      AlphaNet Pty Ltd

      Box Hill, VIC, Australia.
      Tel: +61 3 86771500
      • Prepaid Solutions
      • Wholesale Terminations
      • VoIP Phones and Gateways
      www.alphanet.com.au,
      sales@alphanet.com.au,

      Apphone Pty Ltd (Asterisk.au)

      Sydney, NSW, Australia.
      Tel: +61 2 97994843, Mobile: +61 2 80028015
      • Calling Card Solution (Net-to-phone & Phone-to-phone)
      • Consultation and Sales Complete Solution (SIP,IAX,H323)
      • Roaming Extensions
      • Design and Implementation of Asterisk solutions (Analog,ISDN,E1)
      • A-Z (worldwide) Voip Call termination (SIP, H323, IAX)
      • Support of Asterisk PABXs
      • VoIP Phones and Gateways (Cisco, Quintum & others)
      • Linux
      • Networking
      • Cisco switching & routing
      www.apphone.com

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    • 12/29/15--21:29: Polycom Phones
    • Everything you need to know about Polycom phones, including SoundPoint and SoundStation phones, where to buy, and more can be found on this page.

      SoundPoint and SoundStation VoIP Phones


      List of features of the Polycom phones

      • User can add/change their directory on the phone
      • Config change /ringtone/directory is uploaded to server
      • Configurable of different ringtone on the phone PER line
      • Have a tone warning if call is on hold, or have MWI waiting (configurable)
      • Intercom, be able to set the delay between no ring, ring and auto answer (configurable)
      • The phone can shorten the name of incoming callers, EX Marc Roger Oliver Chouinard, will show M R O Chouinard
      • have the LAMP indicator
      • be able to configure different voicemail server for each line
      • can modify the location of the buttons on the phone
      • you have a lot of dedicated function keys
      • do not disturb feature
      • be able to access quickly directory, miss call, made call... using the arrows
      • you can Dial a number on the phone, and after pickup the handset. You don't have to pickup a line or taking the handset before be able to enter the number on the phone
      • on incoming call, you can refuse the call, so it stops ringing and goes to voicemail (if no other device is available to ring)
      • you can test audio quality of the phone using internal recording system.
      • configuration is EXTREMELY EXTENSIVE, using a XML interface, and uploaded via a FTP or TFTP server (BR 2.6/SIP 1.4) or HTTP, HTTPS, or FTPS server (BR 3.0/SIP 1.5)
      • the phone uploads its log file to the boot server; you can force a logfile upload also
      • Internal switch doesn't reset when rebooting the phone (it keeps its VLAN settings)
      • Have different dialplan for every Line
      • The IP 600/601 supports a XHTML browser and a custom static XHTML idle screen
      • Supports shared lines (but asterisk does not) - Anyone having details on the specifications used for Shared Call / Bridged Line Appearances (SIP-B), Please post details!!
      • SIP and MGCP supported on the IP300, IP500 and IP600

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    • 12/29/15--22:23: Sip Trunking Providers
    • This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

      Country specific pages:


      What Is SIP Trunking?

      Traditional phone calls go over phone lines, but with the rise of the internet, Voice Over IP communication systems give you much more functionality to receive a high volume of calls. This is called SIP (session initiated protocol) trunking. These services are offered by the majority of VoIP providers and can be tailor made to suit the needs of a small business all the way up to a large corporation with a gigantic workforce and everything in between.

      One of the main advantages of going with SIP trunking is that it will integrate everything from data, the internet, your video, and your voice services all into a single line. The term "trunking" is a bit dated and goes back to a time when traditional telephone services were used and it described how a wide variety of telephone users could share a much smaller pool of communication paths, thus making it a much more efficient option for businesses than getting a separate phone line for each worker. That was in the past, though, as today's SIP trunking capabilities will even allow all calls to be considered local calls and won't even require the business to have a single phone line at all.

      The Benefits of Using SIP Trunking Services

      Choosing to use a good SIP trunking service will significantly level the playing field for smaller business to compete with large corporations and appear much more professional and bigger than they actually are. For larger business, one of the main advantages is being able to much more easily maintain that local presence. Regardless of the size of the business itself, SIP trunking's biggest benefit overall is simply the reduction in cost to the company's overall expenses. This is particularly true for SIP VoIP services that are specifically crafted for a business in particular.
      • It offers very low cost calling.
      • It's much easier to scale than other options, making it very future proof.
      • SIP trunking makes moving a breeze, as you can easily take your phone number with you while moving offices or even while traveling.
      • Network outages are much less impactful, as incoming calls can easily be routed to other locations.
      • It's ideal for any sized business with at least 25 physical phones.
      • It's a fantastic choice for any business that has an international location.
      • It offers the same sort of emergency usages and will even offer enhanced 911 dialing options.

      How SIP Trunking Can Take Your Business To The Next Level

      It used to be that you had to have a PRI (Primary Rate Interface) in order to have video, voice, and data all on a single line. This was a physical device that had to be purchased separately but along with your chosen phone provider's calling plans. This is no longer the case with SIP, as it allows your provider to set up a proxy server (commonly referred to as a SIP proxy), which will provide the outgoing, incoming, and PBX calls for you. Everything is located off site and handled by a professional IT team or contractor through your provider, meaning it saves you a ton of time and money on the technical support side of the equation.

      Being temporarily shut down means simply having the SIP reroute calls to a different office and also allows you to much more easily outsource to international locations, such as with customer service centers.

      All of these benefits are much better than what traditional PSTN (public switched telephone network) have been able to provide through their services and will give your business just what it needs to take it to the next level. ...

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    • 12/29/15--22:49: Virtual PBX providers
    • Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.

      What Is a Virtual PBX?

      A PBX, short for private branch exchange, is a telephone system with the capacity to switch calls between different users on local lines while still relying on the same number of external phone lines. With a virtual PBX system, the system is posted and software based without all of the traditional hardware of a physical PBX.

      Virtual PBX Primary Function

      A virtual PBX is used by businesses in a variety of ways. Primarily, companies utilize the system as an auto-attendant to establish preset call transfer options without needing an operator or receptionist. This type of system is capable of performing tasks that include auto-attendant settings, time of day or day of week functions, or even find or follow me sequences.

      One of the most important functions of a virtual PBX system for companies is the software’s ability to establish pre-determined sequences. For example, in some businesses it may be appropriate for the phone to ring to a receptionist or operator first. If the receptionist does not answer in a predetermined number of rings, however, the call is then transferred to a secretary. Again, if the call is unanswered, it can be set to forward to an assistant. Left unanswered by these two individuals, the call can be forward to a manager or even an owner. These call settings are completely customizable and can be based on any number of sequences.

      This type of software is also able to facilitate customized answering menus and sub-menus. The system can be modified to establish appropriate dial prompts leading to a number of different departments within the business, including different sequences on different days. PBXs are used by the vast majority of businesses to establish advanced call routing services.

      Virtual PBX Cost

      A virtual PBX is a complex service; however, that doesn’t mean that it is expensive. In fact, a virtual PBX is typically more cost effective than a physical PBX. The main reason that a virtual system saves on cost is that it does not require the same investment in capital to establish or set-up the call system. Because a virtual PBX is a software or hosted system, it is typically an operational cost, or a low monthly payment rather than a large upfront investment. This aspect alone generally makes a virtual or hosted PBX a less expensive, or at least more cost effective, option compared to the traditional PBX.

      Virtual PBX Benefits

      Aside from offering an effective call system, a virtual PBX presents a number of added benefits for users. As a whole, virtual PBXs lead the industry in business communication choices. This type of system seamlessly integrates the call management system with any existing phones to affordably and effectively deliver better call management. These systems also feature several innovative call features to meet the needs of any business. These systems offer various functions including call routing, follow and find me call forwarding, voicemail notifications, call recording, and more.

      The benefits aren’t limited to the features, though. Virtual PBXs offer virtually limitless application for one or hundreds and even thousands of employees. Likewise, there is not hardware to maintain or constantly upgrade. Considering that benefit, the system is also more cost effective and generally provides for a variety of flexible billing options. The limited maintenance, web-based management, and hassle-free setup alone are often enough to convince a company to switch over to this option.

      PBXs are an important tool in any business that makes and receives nearly any volume of calls. A virtual PBX can dramatically increase the efficiency of a business by effectively managing calls. This efficiency combined with the other numerous benefits of a virtual PBX can virtually transfer the communication capabilities of any company.

      List of Virtual PBX Providers




      OnePipe


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