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Posted August 12, 2011 by in Advertising, Buy Sell Minutes, Free VoIP Software, Free VoIP Zone, Looking for Buy Minutes, Looking for Sell Minutes | No comments yet

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Posted August 10, 2011 by in Advertising, Buy Sell Minutes, Free VoIP Software, Free VoIP Zone, Looking for Buy Minutes, Looking for Sell Minutes, Mobile VoIP | No comments yet

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Skype Remote Calling Using DTMF Recognition

Posted April 3, 2011 by in Free VoIP Software | No comments yet

// Created/Copyright by TheUberOverLord, for public domain use, please don’t remove this line.

// Example of using the Skype public API, Skype4COM, asynchronous TCP sockets and DTMF recognition

// This rough example is meant to be used in debug mode.

// Version 3.0.0.1

// Purpose:

// To demonstrate remote calling from mobiles and landlines or Skype name to Skype name

// using the Skype public API with Skype4COM to process voice data via a Asynchronous TCP

// port interface to call Skype names and telephone numbers with and without speeddial

// numbers using both Skype call transfer methods for inbound calls and Skype conference

// calls for outbound (“Call Me Back”) mode calls as well as playback Skype voicemails

// by using DTMF command recognition methods while providing limited access ability on

// what Skype names or telephone numbers (“PSTN Caller Id’s”) are allowed to use these

// remote calling services using Microsoft C# and no managed threads.

// Features:

// Skype inbound/outbound Skype call processing including:

// Skype to Skype calls

// PSTN to Skype calls

// Skype to PSTN calls

// Skype conference calls using both PSTN to Skype and Skype to Skype

// Skype call transfer

// Voicemail playback in calls

// Skype Speed-Dial Numbers

// Asynchronous TCP port processing using Skype public API methods

// DTMF recognition for both inbound and outbound calls using the Goertzel algorithm

// Same call recovery logic for DTMF commands. No need to hang up and try again

// “Call Me Back” mode

// This example allows you to call your Online Number(s) (“Formally called SkypeIn”)

// or Skype name(s) remotely and make Skype calls using telephone numbers or Skype names

// and telephone numbers using speeddial numbers from other phones or other Skype names.

// It’s “Skype To Go” on Steroids.

// There are many areas of the world where “Skype To Go” is not available yet but

// Skype Online numbers are. This example shows how to create your own extended version

// of “Skype To Go” like features with added features of being able to call thousands of

// Skype names or telephone numbers using Speeddial numbers and any telephone number of

// your choice as well as use a “Call Me Back” mode if it would be cheaper to have a Skype

// conference call, so that your mobile is using a inbound call

// (“Some Mobile plans don’t charge for inbound calls”) than do the same as an outbound call.

// “Call Me Back” mode has some addtional features that all users of this example may choose

// to use as well, and not because of cost.

// NOTE: “Call Me Back” mode uses Skype conference calling, so be aware that if you are

// calling a telephone number, that is using Skype credits, that you will be charged

// normal Skype rates for both the call back to you, as well as the call made to

// the other party.

// This is not an issue if you have a Skype subscription that covers both calls

// or the called party is a Skype name and the phone you called from is covered

// by a subscription. If you are calling a Skype name, and do not have a subscription

// that covers the call back to you, in that case, you would be charged normal Skype

// rates for only the call back to you from Skype.

// Call Transfer (“Which is used in this example for Normal mode processing”) will NOT

// transfer a call to a Skype user that is Offline and that does not have Skype voicemail.

// Even if that Skype user has call forwaring active and even if you have voicemail. Normally

// if you have Skype voicemail and you call another Skype user that does not, you can still

// leave a Skype voicemail. This is not the case when using call transfer.

// So you can use the “Call Me Back” mode, when/if needed to Call a Skype user that

// happens to be offline but does have call forwarding activated. Because “Call Me Back”

// mode uses Skype conference calls to do processing for this example and Skype conference

// calls do support calling a Skype user who is Offline but has call forwarding activated.

// However, Skype conference calls (“Which are used for “Call Me Back”) mode do not support

// creating a conference call with a Skype user who is Offline and does not have call

// forwarding activated.

// So you might be asking yourself, how would you know to use Normal mode or “Call Me Back”

// mode using this example. The answer is that if a call to a Offline Skype name fails

// using this example in either mode you will receive DTMF response tones that will inform

// you of if the other mode could in fact make the call using the other mode

// successfully or not to that Skype name. If the call could not be made successfully in

// either mode for that Skype name, you can then try calling their Home, Mobile or Office

// number, if that information is found in that Skype users profile data.

// While in “Call Me Back” mode there is no need to hang up every time you wish to make

// another call. Once the current call is finished. If you wish, you can wait until the

// current called party you called hangs up and wait about 5 seconds and start a new call,

// I say about 5 seconds, because it can take about that time for the API to process that

// the called party has indeed hung up.

// This example has error-recovery for every situation possible so that you do not need

// to hang up and call back because you entered a bad speeddial number or a bad phone

// number. Instead you will receive DTMF responses to tell you what happened. So that you

// can continue in the same call.

// NOTE: To receive PSTN calls you will need a Online Number (“Formally called SkypeIn”).

// If you don’t have an Online Number, you can still use the example by calling

// a Skype name instead of your Online Numbers. In both cases you can call other Skype

// names using speeddial numbers and/or if you have a Skype subscription or Skype

// credit you can also call telephone numbers using this example as well.

// If you have a Skype subscription calls transferred or created using Skype conference

// calls covered by your Skype subscription are at no cost using this example.

// Calls not covered by a Skype subscription will be at normal Skype rates.

// DTMF commands supported:

// # Playback unread voicemail messages.

// *# Call Me Back – Some Mobile plans don’t charge for inbound calls.

// This allows you to be called back and use the same features.

// 1# This calls speeddial 1. Which could be a Skype name or SkypeOut Contact.

// 1*1# If speeddial number is a Skype name the profile data for that Skype name will be

// looked at and if possible be used to call the Skype name Home number.

// 1*2# If speeddial number is a Skype name the profile data for that Skype name will be

// looked at and if possible be used to call the Skype name Mobile number.

// 1*3# If speeddial number is a Skype name the profile data for that Skype name will be

// looked at and if possible be used to call the Skype name Office number.

// *13126784527# Telephone number with no sequential same digit DTMF = 1 (312) 678-4257

// *3125*52*2*# Telephone number with sequential same digit DTMF = 1 (312) 555-2222.

// NOTE: Processing sequential same digit DTMF is virtually impossible at any

// industry standard because PSTN calls currently use the G729a codec

// which compresses silence and there is no out-of-band DTMF so we need to use

// in-band DTMF. This is why this odd sequential same digit DTMF method is used.

 

// Speeddial numbers can be easily created to avoid any need to remember this odd

// method, if needed. Simply create a SkypeOut contact for that telephone number.

// If any errors are encountered during processing, DTMF is returned and there is

// no need to hang up if for example you entered an invalid speeddial or

// telephone number, or if a Home, Mobile, Office phone was not found.

// DTMF returned codes to the caller:

// 1 Tone response – Completed Voicemail processing.

// 2 tone response – Voicemail processing is not enabled. See enablevoicemail.

// 3 tone response – Speeddial number not found in speeddial list.

// 4 tone response – Invalid speedial index x*(y)# Allowed y values are x*1# = Home, x*2# = Mobile, x*3# = Office.

// 5 tone response – No Home x*1# or no Mobile x*2# or no Office x*3# found for speeddial number in contact profile data.

// 6 tone response – Cannot try to find a Home, Mobile or Office number for a SkypeOut contact.

// 7 tone response – Speeddial x number requested is larger than contact list.

// 8 tone response – Call cannot be completed but using the other mode it could be.

// 9 tone response – Call cannot be completed even using the other mode it would fail.

// 10 tone response – Unable to transfer call, invalid telephone number.

// 11 tone response – Unable to transfer call, Skype denied the call transfer request.

// 12 tone response – Skype4COM exception. You can still continue.

// This can be caused while in Normal mode for invalid telephone

// number entered or not enough Skype credit to make the call if

// the call is not covered already by a subscription.

// 13 tone response – The call was not answered by the distant end.

// 14 tone response – If callbackonfailedconferencecalls is true. Invalid telephone number

// or not enough Skype credit to make the call if the call is not

// covered by a subscription.

// 15 tone response – “Call Me Back” mode – Skype conference calls disabled.

// 17 tone response – Normal mode – Skype Call Transfer disabled.

// 18 tone response – Calls to telephone numbers disabled.

// 19 tone response – Only speeddial numbers allowed. You can still use SkypeOut contacts.

// 20 tone response – Called party call terminated due to onlyallowfreecalls = true.

// There is a switch for testing that allows you to assign speeddial numbers “0″ – “9″

// as authorized users of this example. The default is process all calls for DTMF. See

// answerallcalls.

// NOTE: If you leave answersallcalls set to true any and all inbound calls you have

// Will be auto-answered and be allowed to enter DTMF commands using this example.

// Known Issues fixed in Version 3.0.0.1

// 1. While in “Call Me Back” mode an invalid telephone number will cause the conference

// call attempt to fail. This is because conference calls are used vs

// call transfer for “Call me Back” mode and since there is no way to validate the

// telephone number prior to making the conference call if a bad telephone number

// is used, you will never get called back with a conference call, in those cases.

// Solution: There is a switch called callbackonfailedconferencecalls if set to true

// this example will call back the caller when/if any conference calls fail

// while in “Call Me Back” mode. True is the default.

// How to use this example:

// Simply create a C#.NET Form project use the default name for the project and

// delete the entire contents of Form1.cs Add these contents and right click

// on Your C# project solution and add a reference using the COM tab to Skype4COM,

// build then debug. There are many MessageBox’s that can be uncommented

// to help you test. The lastest version of the Skype4COM.dll can be found here:

// http://developer.skype.com/accessories you need to have it and register the .dll

// NOTE: If you are using Visual Studio 2005, remove System.Linq above

// If your solution/project name is not WindowsFormsApplication1 then change

// namespace below to match your solution/project name.

// If you would like to see what is going on behind the scenes with this example

// you can monitor what is going on between this example and the Skype public API

// by installing SEHE (“Skype4COM Event Handler Example”) as a monitor from here:

// http://www.saveontelephonebills.com/skype/SEHE/ SEHE can be removed at any time

// using your Windows control panel Add/Remove programs. To also learm more about

// SEHE Please go here as well: http://forum.skype.com/index.php?showtopic=142821

// Please go here for any questions about this example:

// http://forum.skype.com/index.php?showtopic=807811

VoIP Emulator

Posted March 8, 2011 by in Free VoIP Software | No comments yet

Megaco (H.248) VoIP Signaling Emulator

VoipEmulator is a MEGACO signaling testing tool, offers developers and QA testers the ability to perform sophisticated Megaco (H.248) signaling functionality testing (Fax, T.38, 3WayCalling, Basic call…).

Easily emulates any Media Gateway Controller (Soft Switch) behavior, thereby increase interoperability with a large scale of VoIP implementations.

http://www.voipemulator.com/

voip switch

Posted January 13, 2011 by in Free VoIP Software | No comments yet

I need original voip switch in monthly rent… who can provide me? plz contact- 01671494208


FREE Callshop Software

Posted September 13, 2010 by in Free VoIP Software | 1 comment

Free Callshop Software for immediate use.

Just register online at http://www.my-hostedcallshop.com and start making calls

Free Phone calls with Gmail

Posted August 26, 2010 by in Free VoIP Software | No comments yet

Google is taking on internet telephone companies like Skype by allowing users to call from its free web-based email service.

The service allows users to make calls to land lines and mobiles from inside their Gmail account.

Phoning anywhere in the US and Canada will be free until the end of the year, while calls to the UK, France, China and Germany will cost 2 cents a minute.

Until now Google offered computer-to-computer voice and video chat services.

“This is a real big deal because now hundreds of millions of Gmail users can make phone calls right from their Gmail page,” Craig Walker, product manager for real-time communications told.

“They don’t need to download an additional application or anything to start making really high-quality low-cost calls. For the user it means much more efficient and low-cost communications.”

The product will initially be rolled out in the US, the firm said. However, for a brief time, international users were also able to use the feature because of an error.

“Unintentionally we briefly made the service available to non-US English users,” a spokesperson said. “We do hope to bring it to our international users soon”

When it rolls out the product link will appear on the left hand of the Gmail page within the “chat” window. A “call phone” option will pop up along with a number pad to let you dial the number of the person you want to talk to.

Google said money raised from international calls will pay for the free US and Canadian calls.

“What surprised me was that they actually said they hope to make money off the calls,” said Danny Sullivan, editor-in-chief of technology blog SearchEngineLand.

“Normally Google is like ‘We don’t know how we are going to make the money’ or ‘We will make money down the way, don’t worry about it’ and this stands out as a big benefit that they get actual revenue early on.”

Competition

Skype, which is the most successful internet phone offering, claims to have over 560 million registered users. The firm said 124 million used the service at least one a month while 8.1 million were paying customers.

The company is planning to offer shares to the public later this year. Observers said that it is too early to say whether companies like Skype should be worried.

“Skype is a well known company in this place and they are almost like a verb in the internet calling world in the way Google is with search. You Skype someone. So I think there is some inertia there to get over and I am interested to see how Gmail users respond,” said Tom Krazit, senior writer with technology news site CNET.com.

“But you always have to worry when Google comes after what you do. They don’t do things half way and bring a lot of resources to any problem they try to tackle. It doesn’t mean you are doomed.

“Google’s product won’t work on your mobile browser so Skype has an advantage there but I don’t think it is a stretch to assume Google will come out with a mobile version pretty soon,” said Mr Krazit.

The company plans an eye catching way to get non-Gmail users to give the product a go. It is in negotiations with a number of university campuses and airports to install red telephone boxes around the country to give users the chance to dial and try.

click here to Download & start calling

Good luck

QuteCom: free voip softphone.

Posted May 23, 2010 by in Free VoIP Software | No comments yet

QuteCom is the new name for the open source softphone previously known as WengoPhone. qutecom is a SIP phone which allows users to speak at no cost from one’s computer to other users of SIP compliant VoIP application. It also allows users to call landlines, cellphones, send SMS messages & to make video calls. None of this functionality is tied to a specific SIP provider & can be used with any provider available on the market, unlike proprietary solutions such as Skype.

General

codec support is based on Verona. for more info
please visit

Audio

Name
Status
Rate (kHz)
bitrate (kbps)
G729 yes* 8 8
G711 yes 8 64
iLBC yes 8 15
AMR-NB yes* 16 5-12
G722 yes 16 64
Speex yes 8/16 4-44
AMR-WB(G722.2)* yes* 16 7-24

* license needed

Video

Name
Status
bitrate (kbps)
H263 yes ?
H263+ no ?
H264 no ?
Dirac* no ?

Download QuteCom RC3 installer for Windows
Download

QuteCom RC2 installer for Mac/OSX
Download

Vhani.com ( A large agent of outsourcing Company )

Posted May 22, 2010 by in Advertising, Buy Sell Minutes, Developer Tools, Free VoIP Software, Free VoIP Zone, Hardware Buying Offer, Hardware Selling Offer, Hosted PBX, Hosted Platform, Looking for Buy Minutes, Looking for Sell Minutes, Mobile Phone, Mobile VoIP, Mobile VoIP Provider, Mobile VoIP Software, Residential VoIP Provider, Sell Buy Hardware, Sell Buy Software, Soft Switch, Software Buying Offer, Software Selling Offer, VoIP Directory | 2 comments

Vhani.com is a Value Added Agent web, a leading IT,Garments,Real Estate (BD), Footwear marketing and sales of outsourcing company.

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Domain Registration
Hosting
Web & Software Development
Clipping path
Data Entry
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Online Techno Shop,
Estate Agents-UK [[ Sales, Lettings, Mortgage Arrangements,Home Renovation,Property Management ]]
Real Estate (BD),
Situdent’s Embassy
Media (Natok & Filim)
Vhani Foundation
Investment in BD

Dhaka-Bangladesh Office:

Mr, Tanvir Mohammad Dipu

Executive –Director

602, Amin Court ,62/63 Motijheel, Dhaka-1000, Bangladesh

E-mail: info@vhani.com

Web: www.vhani.com

London-UK Office:

Md.Khairul Islam (Ashiqbabu)

Director-Communication

Vhani.com

Mobile Number: 07574915715 / 07424081515

Overseas Call:  +00447574915715 / +00447424081515

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Or

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E-mail: info@vhani.com

Web: www.vhani.com

Ekiga ~ Free Your Speech

Posted May 21, 2010 by in Free VoIP Software | 1 comment

Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant
Messenger application over the Internet. It supports
HD sound quality and video up to DVD size
and quality. It is
interoperable
with many other standard compliant softwares, hardwares and service
providers as it uses both the major telephony standards
(SIP and H.323).

Main Features of the Ekiga Softphone in a nutshell

  • Ease of use with a modern Graphical User Interface.
  • Audio and Video free calls through the internet.
  • Free Instant Messaging through the internet with Presence support.
  • Audio (and video) calls to landlines and cell phones with support to the cheapest service providers.
  • High Definition Sound (wideband) and Video Quality up to DVD quality
    (high framerate, state of the art quality codec and frame size).
  • Free of choice of the service provider.
  • SMS to cell phones if the service provider supports it (like the default provider).
  • Standard Telephony features support like Call Hold, Call Transfer, Call Forwarding, DTMF.
  • Remote and Local Address Book support: Remote Address Book support
    with authentification using the standard LDAP technology, Local Address
    support in Gnome (Evolution).
  • Multi platform: Windows and GNU/Linux
  • Wide interoperability: Ekiga use the main deployed stantards for
    telephony protocols (SIP and H.323) and has been tested with a wide
    range of softphones, hardphones, PBX and service providers.

Would you like to
download Ekiga?

Detailed features

Graphical User Interface

  • HIG Compliant GUI: Ekiga looks right, behaves properly, and fits into the GNOME user interface as a whole.
  • Translated into Many Languages
  • Contact centric interface: preferred contacts can be listed in the main user interface (Buddy List).
  • Contact’s presence display: Offline, Away, Do Not disturb, Online status (SIP/SIMPLE).
  • Automatic addition of network neighbours contacts in the main user interface (using the Bonjour/ZeroConf protocol).
  • Optional call panel.
  • Tray icon with call-in notification.
  • Connection quality meter in the statusbar.
  • Message Waiting Indications Support: Ekiga can tell you how much Voice Mails are waiting.
  • Easy to use short-cuts.
  • Easy account selection when dialing for a contact: auto-completion .

Call

  • Call Hold: This effectively pauses Video & Audio transmission.
  • Call Transfer: You can transfer the remote user to another user.
  • Call Forwarding on No Answer, on Busy, on Always: This allows you
    to configure Ekiga to forward incoming calls to a specified user.
  • Message Waiting Indications Support
  • Dialpad: This allow you to dial numbers using a graphical alphanumeric keypad.
  • DTMFs Support: This feature is necessary when using services asking you to dial numbers.
  • Calls History: This is a convenient way about all outgoing and incoming calls.
  • Call Monitoring: Statistics about the network traffic caused by Ekiga are displayed in the status bar.
  • Enum Support: Enum is a standard method to provide a unified
    numbering system between the public switched telephone network (PSTN)
    and various VoIP providers. It is officially deployed country wide.

Audio

  • Wideband Codec Support: HD sound quality
  • Dynamic Threshold Algorithm for Silence Detection
  • Echo Cancellation
  • Dynamic Jitter Buffer
  • 10 audio codecs supported; including G.711 for high
    interoperability, the best free speech codec (Speex) and HD sound
    (G.722 and Speex Wideband)

Video

  • High Framerate support (up to 30 fps)
  • Configurable Resolution, up to DVD quality (from 176×144 to 704×576)
  • Full-Screen Video
  • Support for hardware rendering (DirectX under Windows, XVideo under GNU/Linux)
  • Quality of images Versus Speed (framerate) slider
  • 6 video codecs supported; including the best free codec (Theora) and state of art video codec (H.264)

Text chat

  • Instant Messaging with built-in smiley support (SIP)
  • Graphical selection of smiley for easy inclusion in text message
  • Presence display: display your peer chat status

Contacts

  • Presence Status with Custom Message (SIP): The status of your
    contacts are displayed in the main User Interface with a custom
    message, and you can set your own Status (Online, Away, Do not disturb)
    with a custom message
  • Advanced Addressbook: The Address Book is a feature which allows
    you to find users to call and/or to save locally your list of persons
    that you call on a regular basis.
  • Remote Address Book Support: Ekiga can load the list of users from a remote LDAP directory (with authentification)

Devices

  • Hotplug: Automatic detection of hotplugging of audio devices and video devices on Linux (ALSA and V4L1/2)
  • Devices Auto-Detection
  • Sound API: on Windows DirectX, on GNU/Linux OSS and ALSA Compatible Soundcards Support
  • Video API: on Windows DirectX 9 video input and output, on GNU/Linux Video4Linux, Video4Linux 2 and Firewire Cameras Support

Voice Over IP services

  • Possibility to Simultaneously Register to Several Accounts: You can
    register as many SIP or H.323 provider accounts as you want, and you’re
    able to use them simultaneously.
  • SIP Compliant: You can use any SIP compliant VoIP provider. They
    may provide you address for VoIP similar to e-mail address, PC-to-Phone
    calls, Phone-to PC calls, Voice mails, …
  • H.323v4 Compliant: You can use any H323v4 compliant VoIP provider. (Gatekeeper -RAS- Support)
  • Outbound Proxy Support: Some providers use a relay for your communications and require this setting.
  • SIP dialog-info notifications: they allow displaying notifications
    of incoming calls in the roster, and being informed of incoming calls
    reaching your contacts (if the server supports it, e.g. Ekiga.net,
    Kamailio and Asterisk do).

Codecs Features

  • SIP Capabilities exchange: Ekiga will automatically select the common codecs between the peers.
  • Video Bandwidth Limitation
  • Intel IPP Codecs
  • Plug-in support for audio and video codecs

Audio Codecs

  • G.711-Alaw
  • G.711-uLaw
  • Speex (NarrowBand and WideBand)
  • G.722 (wideband)
  • iLBC
  • GSM-06.10
  • MS-GSM
  • G.726
  • G.721
  • CELT ultra-low delay (32 kHz or 48 kHz). Experimental

Video Codecs

  • THEORA Video Codec (SIP only)
  • H.264 Video Codec (SIP only)
  • H.263 Video Codec (SIP only)
  • H.263+ Video Codec (SIP only)
  • H.261 Video Codec (SIP and H323)
  • MPEG4 Video Codec (SIP only)

Network

  • Local network integration using the Rendez-Vous/Bonjour/ZeroConf
    protocol: Ekiga users in the local network will show up automatically.
  • Detection of dynamic IP address changes and dynamic addition and removal of network interfaces
  • Assisted NAT Support (automated STUN support): Ekiga has extensive
    and improved NAT support. In most of the cases, you do not have any
    configuration to do.
  • Gateway/Proxy Support

Integration

  • Part of the Gnome Desktop
  • Integration with Novell Evolution 2.00: You can share your contacts with the groupware client for Linux, “Novell® Evolution™”.
  • KDE and GNOME Compatibility
  • Remote LDAP server integration for network address-book integration
  • State-of-the-art LDAP support (supports authentication)
  • Interoperability with PBX like Asterisk™

Configuration

  • Configuration Assistant: The Configuration Assistant is a 8
    step-by-step questionnaire that will guide you through all the steps
    involved in creating the basic configuration you will need to operate
    Ekiga.
  • Configurable Sound Events: You can customize sounds and select a
    special sound device for them to play; you’re able to have the ring in
    your Hi-Fi hardware and the call in the headset.
  • In call Instant-Apply Support for Settings
  • Gnome’s GConf and External Configuration support: Ekiga can store
    its settings in Gnome’s GConf application or in a standalone file (e.g.
    on Windows).
  • Configurable Port Ranges (SIP and H323): Ekiga uses standards
    ports; in rare cases you might change the ports. (Advanced setting)

Protocols Support

Session Initiation Protocol (SIP)

  • SIP re-INVITE support
  • Unique port : Use only one port for all outgoing SIP requests going to the same destination
  • Proxy support
  • Outbound Proxy support
  • SIP: SIMPLE presence support
  • SIP Presence subscriptions : SIP SUBSCRIBE/NOTIFY
  • SIP Presence publishing : SIP PUBLISH
  • SIP Presence document : SIP PIDF
  • RFC2833 DTMFs support
  • SIP INFO DTMF support
  • SIP dialog-info notifications: they allow displaying notifications
    of incoming calls in the roster, and being informed of incoming calls
    reaching your contacts (if the server supports it, e.g. Kamailio and
    Asterisk do).
  • SIP RFCs in the OPAL stack (Ekiga do not use all of them, especially T.38 FAX features).

H.323

  • Registrar Support: You can register to SIP compliant VoIP providers.
  • Gatekeeper (RAS) Support: You can register a VoIP service using H.323.
  • Gateway/Proxy Support
  • H.450.1 Call Hold
  • H.450.2 Call Transfer
  • H.450.3 Call Forwarding on No Answer, on Busy, Always
  • RFC2833, Q.931, and Inband DTMF support
  • H.235 Annex D. Support: Security of calls.
  • H.245 Tunneling and Fast Start
  • H.245 Text Chat during calls (H323)
  • H.323 standards in the OPAL stack.

Windows Features

  • Support for DirectX 9 Video Capture/Output
  • Installer

Administrators

  • Possibility for Administrators to Block Some Settings

Developers

  • GUI and Engine separation to allow GUI customisation/port to other toolkits and integration in other projects.
  • Use of sigc++ in a signal-based organization.

Experimental features

  • Significant improvements in IPv6 support.
  • Gstreamer audio and video capture support.
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BD TDM -0.0350 ,BD IGW -0.0355 ,INDIA CLI 0.0105, INDIA E1-0.0100 ,

by khan on April 13, 2011 - 12 Comments

we sale only quality route . 100% pure . we dont mixd we provide reseller all over the world contact Mr Khan itskhan41@yahoo.com power@khan.com...

Exciting Rates FOR BD||PAK||SRILANKA||INDIA||NEPAL Routes

by david@eaglestate on March 9, 2011 - 5 Comments

Hello Dear Provider, Eagle State International Ltd is a fast developing prominent VoIP service solution provider located in Hongkong. If you have direct qua...

Start making money from international premium numbers!

by pncall on April 2, 2011 - 3 Comments

Hello there, *Can you generate traffic to international premium rate numbers? *Are you looking for fast and secure payouts? *Would you like your own de...

Bangladesh White Route, Coming Eid Sell Offer

by onirbaan on November 7, 2011 - 3 Comments

Bangladesh White Route, Coming Eid Sell Offer Please contact before Coming Eid BTCL we have expanded capacity. (Paid)Test 500 ports paying only 100 $ . We...

Vhani.com ( A large agent of outsourcing Company )

by Md.Khairul Islam Ashiqbabu on May 22, 2010 - 2 Comments

Vhani.com is a Value Added Agent web, a leading IT,Garments,Real Estate (BD), Footwear marketing and sales of outsourcing company. Business & Products:...

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