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  • 09/08/15--00:54: Open Source VOIP Software
  • Open Source VOIP applications, both clients and servers.

    Open source means all source code is available!! Do not post any "free but not open" software here!

    SIP Proxies


    • JAIN-SIP Proxy
    • Mini-SIP-Proxy A very tiny perl POE based SIP proxy
    • MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
    • MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
    • NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
    • Net-SIP A Perl SIP framework that includes a stateless proxy
    • OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. ...

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  • 09/08/15--01:05: VOIP Software
  • Billing

    See Open Source Billing Systems& VOIP Billing

    Call center monitoring


    Computer Telephony Integration (CTI)


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  • 09/08/15--01:11: 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.

    September 2015


    August 2015

    • 2015-08-20 -

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  • 09/08/15--01:28: Codec Software
  • Sources for Codec Software


    From a comp.dsp newsgroup thread
    Someone said that the book: "C Algorithms for Real-time DSP" by Paul Embree includes code
    for g.711 and g.722.

    There is some GSM and G.722 ADPCM code in the Digital Signal Processing Applications Using the ADSP-2100 Family book - volume 2 ftp://ftp.analog.com/pub/dsp/21xx/pre-8x/examples/apps2.zip


    See Also:



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  • 09/08/15--01:45: NAT and VOIP
  • What is NAT?

    NAT (Network Address Translation) is a technology most commonly used by firewalls and routers to allow multiple devices on a LAN with 'private' IP addresses to share a single public IP address. A private IP address is an address, which can only be addressed from within the LAN, but not from the Internet outside the LAN. In order to let a device with a private IP address communicate with other devices on the Internet, there needs to be a translation between private and public IP addresses at the point where the LAN connects to the Internet, that is within the firewall/router connecting the LAN to the Internet. Such a translation is commonly referred to as NAT (for Network Address Translation) and a router doing such translation is often called a NAT router or NAT firewall/router. Sometimes NAT is also called IP Masquerading. The passing of traffic through NAT is called NAT Traversal.

    The way NAT works is in principle rather simple. When a device on the LAN initiates a connection with a device on the Internet, the device will send all traffic to the NAT router first. The NAT router then replaces the source address, which is the device's private address, with its own public address before passing the traffic to its destination on the Internet. When a response is received, the NAT router searches its translation tables to find the original source address of the packet from which the device on the LAN originally started the connection and thus passes the response to that device.

    Unfortunately, when a connection is originated by a device on the Internet outside the LAN it is not clear which device on the LAN the connection is meant to be established with. In this case there needs to be some rule that tells the NAT router what to do with the incoming traffic, otherwise it will simply discard the traffic and no connection will be established. If the NAT router supports what is commonly referred to as a 'software DMZ' it can handle simple rules, such as "pass all incoming connection requests to the device with address 192.168.0.2". Another technique, called port forwarding allows the NAT router to pass incoming connection requests to different devices on the LAN depending on the type of connection (ie web or mail connection). However, if there are multiple devices on the LAN to which a certain type of connection from outside may need to be established, then neither a software DMZ nor port forwarding will be sufficient.

    Sometimes people (those without network experience) have difficult to understand if their host is or not behind NAT, there is a website that will test to see if you are behind NAT (you need to have Java): (amibehindnat.com).


    The Trouble with NAT and VOIP

    In addition, the way in which conventional VoIP protocols are designed is also posing a problem to VoIP traffic passing through NAT. Conventional VoIP protocols only deal with the signalling of a telephone connection. The audio traffic is handled by another protocol and to make matters worse, the port on which the audio traffic is sent is random. The NAT router may be able to handle the signalling traffic, but it has no way of knowing that the audio traffic is related to the signalling and should hence be passed to the same device the signalling traffic is passed to. As a result, the audio traffic is not translated properly between the address spaces.

    At first, for both the calling and the called party everything will appear just fine. The called party will see the calling party's Caller ID and the telephone will ring while the calling party will hear a ringing feedback tone at the other end. When the called party picks up the telephone, both the ringing and the associated ringing feedback tone at the other end will stop as one would expect. ...

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  • 09/08/15--02:16: Pakistan
  • Overview

    The Pakistan Telecommunication Authority regulates all phone, cellular, VOIP, and Internet communications in Pakistan.

    VOIP

    Currently it appears that the PTA offically allows only extermely limited use of VOIP. The authority are highly controlling their network for the couple of years.

    Resources

    • TelecomPK Blog - A Blog on the state of Telecommunication in Pakistan

    See Also


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  • 09/08/15--02:40: Asterisk news and blogs
  • This page is a collection of Asterisk-based news sites and blogs.


    Daily news


    Blogs on Asterisk


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  • 09/08/15--02:51: VOIP Security
  • VOIP Security Issues:

    • Interception of calls
    • Denial of Service Attacks
    • Theft of Service
    • Exfiltration of data via media session
    • Malware embedded in signaling and media session

    Interception of Calls

    VOIP phone calls are fairly easy to capture and decode if you one has physical access to a LAN segment that the VOIP packets travel accross. Fortunately, with most enterprises using Ethernet switches instead of hubs, there are a limited number of locations this is possible.

    Countermeasures
    • Physical Security
    • Encryption - not yet widely available for VOIP services
    • Secure wireless networks

    Denial of Service Attacks

    Sending spurious traffic to VOIP services or endpoints to disrupt normal service.

    Countermeasure
    • Some Session Border Controllers have DoS countermeasures built in.


    Theft of Service


    Countermeasures
    • Use Authentication features of VOIP protocols
    • Encryption
    • Physical security
    • Secure wireless networks

    Exfiltration of data via media session

    Sending data out via the media session. RTP as a covert communication channel.

    Countermeasure
    • Deep Packet Inspection of all outgoing media streams

    Malware embedded in signaling and media session

    Malformed SIP and RTP (or other signaling/media streams) with malicious payloads

    Countermeasures
    • Deep Packet Inspection of all incoming signaling and media streams

    VoIP and Unified Communications Security

    http://ucsecurity.wordpress.com - All about Cisco UC Security. Your one step guide to building, designing, and maintaining secure Cisco UC solutions.

    VoIP Security Forums


    VoIP Security Training


    See Also:


    • SecAst (Asterisk Intrusion Detection and Prevention) SecAst is an Asterisk specific intrusion detection and prevention package, designed to secure any Asterisk server using a range of techniques. SecAst is available in free and commercial versions. More information available at www.generationd.com
    • Securing Internet Telephony: Encrypting Voice with VoIP-over-VPN Ever wonder who eavesdrops on your VoIP conversations? Unencrypted VoIP compromises information security for companies that handle sensitive information and the carriers that serve them. This Patton Electronics white paper explains how you can make your Internet Telephony solution completely secure. Find out why VoIP-over-VPN technology is more expedient than emerging CODEC-based approaches such as SRTP and SIP TLS. You'll also learn how Internet Key Exchange (IKE) simplifies VoIP installation at the same time it strengthens information security
    • easySysAdmin easySysAdmin is an automated support/security platform, designed to save your engineer's time and prevent hacking attempts. ...

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  • 09/08/15--10:16: DID
  • Direct Inward Dialing Number (Also known as DID or DDI)


    DID (DDI) Background

    Most businesses have several incoming telephone numbers used for specific purposes. For example customer service, sales, etc. Some have an individual telephone number for each user in the system. In a home setting on the other hand, each telephone number comes in on a different pair of wires typically. This is not practical in a business environment that has many telephone numbers.

    Why was DID actually Created?

    So DID ("direct inward dialing") was invented as a way to re-use a limited number of physical phone lines to handle calls to different published numbers. In a business with DID, the phone company uses DID signalling to identify the number they are about to connect to the business's PBX. Historically, this was done by pulsing the last 3 or 4 digits of the number being dialed before connecting the number. The PBX would use these DID digits to switch the call to the right recipient.

    In modern PBX's, typically, digital methods (example: PRI) are used to do the same thing, ie. supply the "called party" information. But many business's still have old PBX's which use the analog signalling I mentioned before. The type of telephone lines used for analog DID are different than regular home telephone lines. Usually, battery voltage is supplied by the business PBX instead of the telco. Also, the telco signals a new call by bridging the line instead of by ringing the line. The receiving PBX signals back that it's ready to take the call by momentarily reversing polarity of the voltage on the line (this is called "winking" the line)

    Old Fashion Way: (PSTN WORLD)

    Direct Inward Dialing is used when your PBX telco connection allows direct dialing to extensions within a PBX, using physical lines (or channels on a PRI) on a shared basis. DID service consists of identifying the "called party" by using DTMF or by digital means, before connecting each call. The service can be sent over an E&M Wink T-1 as DTMF and also as D-Channel information on a PRI.

    On a PRI connection, the telco can send only the digits that differ between the group number and the extension (often four digits) or the whole number - it depends on the connection to the telco.

    DID (DDI) in the new VOIP World

    Let's say you buy a phone line from Vonage or some other phone service provider who offers phone service over broadband. The number that they provide to you, in technical terms is a DID number. This is the number that they have assigned to you to connect you to the old PSTN Networks around the world. Any service provider who wants to offer a phone service over IP address, needs to buy DID numbers from his CLEC or any other large service provider like Level 3 in the United States or go to a consortium (company that will take large blocks from many providers and hand them out one at a time)

    If you are using an IP PBX like Asterisk, and you want to connect yourself to PSTN so people can call your office, you can either

    1) Buy an Analog or E1/T1 card from Digium, OpenVox, Rhino Equipment Corp or Sangoma

    2) Buy DID number from DID service provider

    DID Service Providers, convert the analog to digital and provide these DID numbers over the Internet, with SIP or IAX2.

    Service providers such as
    Wholesale DID Numbers, BuyDIDNumber Fax ,Asterisk Supported BuyVirtualNumber

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  • 09/08/15--18:53: 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.

    373K REAL Wholesale.

    • Local/LD Origination - $0.25/DID and No MOU or port charges--free inbound!
    • US and Canada Termination - Over 70% for less than $0.005, no commitments, or port charges/limits!
    • Toll-Free Origination - FREE Toll-Free DIDs and usage rates starting at $0.0002!
    • Toll-Free Termination - Get compensated for your toll-free termination.
    • Support - Live humans answering the phone 24/7. Engineers available for free assistance.
    • There's a reason service providers entrust the traffic of over 60 million users to us everyday.



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  • 09/08/15--22:56: Bandwidth consumption
  • VOIP Bandwidth consumption naturally depends on the codec used.

    When calculating bandwidth, one can't assume that every channel is used all the time. Normal conversation includes a lot of silence, which often means no packets are sent at all. So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet (which adds overhead), the full bandwidth is not used at all times.

    A codec that sends a 64kb stream results in a much larger IP network stream. The main cause of the extra bandwidth usage is IP and UDP headers. VoIP sends small packets and so, many times, the headers are actually much larger than the data part of the packet.

    IAX2 trunking helps with the IP overhead, but only when you are sending more than 2 or so calls between the same Asterisk servers. John Todd has done some useful practical testing, named IAX2 trunking: codec bandwidth comparison notes and results.

    The bandwidth used depends also on the datalink (layer2) protocols. Several things influence the bandwidth used, payload size, ATM cell headers, VPN headers, use of header compression and IAX2 Trunked. You can see the influence of some of this factors using the Asteriskguide bandwidth calculator.

    Teracall has the table which shows how the codec's theoretical bandwidth usage expands with UDP/IP headers:

    Codec BR NEB
    G.711 64 Kbps 87.2 Kbps
    G.729 8 Kbps 31.2 Kbps
    G.723.1 6.4 Kbps 21.9 Kbps
    G.723.1 5.3 Kbps 20.8 Kbps
    G.726 32 Kbps 55.2 Kbps
    G.726 24 Kbps 47.2 Kbps
    G.728 16 Kbps 31.5 Kbps
    iLBC 15 Kbps 27.7 Kbps

    BR = Bit rate
    NEB = Nominal Ethernet Bandwidth (one direction)


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  • 09/08/15--23:14: 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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  • 09/09/15--00:54: Dcap in Iran
  • The list of Certified Asterisk professionals in IRAN


    Asia / Iran


    Omid Mohajerani
    E-mail: omid dot mohajerani at gmail.com
    VOIP Expert ( CCNA - CCNA Voice - Dcap )
    VOIP/CallCenter Consultancy , Training , Installation
    Intelligent IVR Programming
    HA CallCenter Solutions
    A2billing
    CRM Integration with Asterisk
    [http://linux-notes.blogfa.com

    Mojtaba Esfandiari.S
    E-mail: mespio at gmail dot com
    Senior VoIP Consultant
    Solutions Architect
    Specialties at: IP Telephony, Kamailio, Opensip, Asterisk PBX, Call Center,
    VoIP Programing, Development, Monitoring, Trunking,
    Termination, Billing, VoIP Training,
    Developing SIP Protocol Security, Python programmer

    Sajad Sabri
    E-mail: sabri at voip98 dot com
    Expert and consultant in communication solutions
    Specialties at: IP Telephony, Asterisk and Elastix PBX, Call Center,
    VoIP Programming, Development, Monitoring, Trunking, Training,

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  • 09/09/15--04:17: VOIP Billing
  • Hosted Billing Services (in Alphabetical Order)


    • Incorpus TeleNetworks Incorpus provides Class 5 and Class 6 softswitches on very affordable monthly rental plans. These are carrier grade switches suitable for Big Enterprises as well as small companies and individuals who are trying to build their own voip company. All switches comes with strong firewalls and bandwidth optimizer and the plans start from as low as just 80$ monthly.Please visit our website and have a live chat with our sales team for any guidance you need. email us for more information at sales@incorpus.in or info@incorpus.in
    • Zoom Soft's dedicated easy VoIP billing server provides services for VoIP solution from small to cluster solution for handling thousands of concurrent solution. Our VoIPSwitch is class 5 softswitch which is integrated with billing VoIP Switch. This system also integrated with high features and facilities. Our billing server started with only $50 monthly with 24/7 support. Directly order could be possible with our VoIP billing portal system.
    • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
      • Note:CloudAstrix SPE Module works with FreeSwitch.
    • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
    • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
    • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
    • Adore All-in-One SIP Server and Client v2.2.1 - new released with Class5 features
    • Aradial AAA for Billing Solutions
    • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. ...

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  • 09/09/15--09:02: Asterisk consultants USA
  • This is a comprehensive list of Asterisk consultants in the USA (United States). Add your entry here (alphabetical order, by state and company), but stick to states where you have actual presence!

    Feel free to add a few lines (max 5) describing your business. Don't forget to add VoIP telephone numbers, like a SIP URI. Use common courtesy with others' entries! No images!


    ALABAMA


    Asteria Solutions Group


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  • 09/09/15--11:07: IVR
  • What Is IVR?


    IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

    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 key-presses 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 touchtone 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.

    Here is an example of IVR implementation in Voice broadcasting

    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
    • IVR System Simulation Model - estimates resources required for an inbound calling campaign.
    • IVRS World - Blog about IVR

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  • 09/09/15--16:55: Asterisk AEL2
  • Asterisk Extension Language v.2

    AEL v.2 is intended to provide an actual programming language that can be used to write an Asterisk dialplan. It
    further extends AEL, and provides more flexible syntax, better error messages, and some missing functionality.
    AEL v.2 is a new version of the AEL compiler, written by Steve Murphy. It was originally introduced as a large asterisk patch in the Asterisk bug database. See: Bug 6021.

    AEL is really the merger of 4 different 'languages', or syntaxes:
    • The first and most obvious is the AEL v.2 syntax itself. A BNF is provided near the end of this document.
    • The second syntax is the Expression Syntax, which is normally handled by Asterisk extension engine, as expressions enclosed in $[...]. The right hand side of assignments are wrapped in $[ ... ] by AEL, and so are the if and while expressions, among others.
    • The third syntax is the Variable Reference Syntax, the stuff enclosed in ${..} curly braces. It's a bit more involved than just putting a variable name in there. You can include one of dozens of 'functions', and their arguments, and there are even some string manipulation notation in there.
    • The last syntax that underlies AEL/AEL2, and is not used directly in AEL/AEL2, is the Extension Language Syntax. The extension language is what you see in

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    Company Profile:
    Ecosmob Technologies Pvt. Ltd. (commonly as known, Ecosmob) is India’s leading IT Company offering various IT software solutions and services. It was introduced in 2007 to provide complete IT based solutions and services. It has its headquarter in Ahmedabad, Gujarat. It has been delivering flexible, simple and affordable IT solutions to the renowned enterprises overseas.

    Ecosmob Technologies has secured a leading place in the VoIP industry with its next generation VoIP solutions and products. The team has the team of experienced developers who have rich expertise in developing customized VoIP software based on the client requirement. The company also offers open source consulting services. In nutshell, the company provides the design, development, deployment, consultancy and support services in the VoIP technologies such as:

    • WebRTC
    • Asterisk
    • FreeSWITCH
    • Kamaliio
    • OpenSIPS
    • OpenSERs

    We provide custom design, development and deployment services for VoIP solutions. Some of them are briefed below:

    Conferencing Solution:
    The company provides a comprehensive conferencing software solution to conduct voice, video and web conferences. It provides custom development services for conferencing solution with selected features from the whole range of features the company offers including, Personalized meeting rooms, Web Phone, Conference wise Polling, Live Conference Viewer, and more. Being an environment friendly solution, the conferencing system is not just reducing the corporate carbon footprints, but it saves the travelling costs and the time to schedule the meeting without any location constraints.

    IP PBX Solution:
    Custom IP PBX solution allows media communication to take place with the help of a PBX combined with VoIP. The IP PBX software solution is not just limited to call features, but also includes an interactive directory listing, DND, conference bridging, IVR, privacy management feature, etc. Being a web/GUI based configuration, the IP PBX system eliminates the phone wiring and vendor lock-in that results in offering better customer productivity and services.

    Hosted PBX Solution:
    The company produces the Hosted PBX software solution to administer the communication that takes place without the need of any hardware. The development services that the company provides for Hosted PBX solution includes auto attendant, caller IDs, customized message alerts, fax to email, voicemail, find me, follow me, call waiting, conferencing and forwarding as their key features. Improved customer support, reduced cost of both phone bills and hardware are the added advantages to this system.

    To get more details about us or to discuss your specific requirements, contact us:
    Website:https://www.ecosmob.com
    Phone: +1 303 997 3139 or +91 079 40054019
    Email:sales@ecosmob.com

    Stay connected with us on social media!
    LinkedIn:https://www.linkedin.com/company/ecosmob
    Google+:https://plus.google.com/+ecosmob
    Twitter:

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  • 09/10/15--07:12: BULK SMS
  • Bulk SMS, also known as bulk messaging or bulk text messaging is the act of disseminating SMS messages in large numbers so they are able to be delivered to various mobile phone terminals. This form of messaging is typically utilized by consumer brands, banks, enterprises and media companies. The messages are generally used for mobile marketing, enterprise and entertainment. However, banks often use them for fraud control. For example, if criminals are circulating a fake email that is asking people who have accounts at a certain bank to provide their social security numbers or other confidential information, these text messages can alert people to the scam so they do not fall victim to it. Bulk messaging is often utilized for reminders and alerts. However, it is more frequently used to send communications and information between customers and staff of various companies. Bulk messaging enables the delivery of SMS messages to large numbers of mobile phones that are located all around the world.

    Bulk messaging software

    In order to receive and send bulk messages, software is needed. There are many types of software packages specifically designed for this task that are available. These packages give their users the ability to send messages to as many phone numbers as they want. There are many different ways in which these phone numbers can be managed.

    The vast majority of software applications that are designed to be used with SMS enable the user to upload mobile phone number lists with the use of a CSV or TXT file. Systems that are more advanced are capable of automatically deleting any numbers that are repeated. There are also systems that can be programed to validate all of the mobile phone numbers before the messages are sent to them.

    Enhanced software features are also currently available that allow users to schedule messages to be delivered at certain days and/or times. Bulk messages are also able to be sent on mobile networks that are international or national, assuming that the provider of the bulk messaging software sends internationally.

    Bulk messaging portal

    Bulk messaging features can be added to websites through the use of this specific online script. Unlimited mobile phone numbers can be added to the list of numbers to send messages to. There are a wide variety of ways that can be used to manage these numbers.

    Bulk messaging API

    The majority of services that handle bulk messaging use the API's (Application Programming Interface) listed below. These enable the addition of functionality to programs by their programmers:

    • Email
    • HTTP
    • SMPP (Short Message Peer to Peer)
    • FTP (File Transfer Protocol)

    Immediate benefits of using bulk SMS messaging

    When a particular business is not doing well financially, they need to utilize various tools that can help them gain a competitive advantage in their specific industry. One of the main reasons that bulk SMS is so popular is its ability to lower operational costs while also generating revenue at the same time. Bulk SMS might be the only medium that is able to show a return on investment that is able to be measured. Wholesale SMS messaging is targeted, which makes it extremely effective at getting people to respond and generate revenue.

    Reduces operational costs

    Bulk SMS message transmission is more effective than email and less expensive than voice calls. There are thousands of businesses located all around the globe that utilize wholesale SMS as a way to communicate with their suppliers, employees and customers. There is a significant cost savings as a result of time being saved because actual voice calls to suppliers, employees and customers do not need to be made. A single message can instantly be sent to many people at the same time, as long as the person is located in an area with mobile coverage. The ability to disseminate information so quickly to a large target audience reduces communication costs while also generating revenue if used for marketing purposes.

    Allows customers to be accessed easily

    More people have mobile phones than have access to email or landline phones. Every mobile phone supports the use of text messaging. All mobile phone users are comfortable using this technology because it is simple and easy to understand. This makes wholesale SMS the perfect medium to use for communication with customers. There are also no demographical or geographical restrictions. ...

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  • 09/10/15--07:24: voip-info.org
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