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  • 11/02/15--05:23: VOIP Event Calendar
  • 2015 VOIP Related Event:

    November 2015

    • 09 – 10 OpenSIPS Summit 2015 in Austin, TX - Come and help us "Keep Austin Wierd" at our eclectic venue, The Contemporary Austin Jones Center. We'll be presenting, via presentations and workshops, all the great additions we have in OpenSIPS 2.1 - webRTC, async queries, SIP compression, fraud detection and many others. The Summit is not only about 2.1, but also about tools, platforms or engines which are OpenSIPS interconnected (like billing, interfaces and others). We'll also have presentations from companies using OpenSIPS in their production environments, giving us insight into it's real world applications.
    • 11 – 13 OpenSIPS LIVE BOOTCAMP 2015 in Austin, TX - Enjoy a rare opportunity to train with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! As part of OpenSIPS week in Austin, we'll be holding a LIVE training following the conclusion of the OpenSIPS Summit on Nov 10th. If you've ever wanted to become an OpenSIPS Certified Professional, this is you're chance! You'll have the rare opportunity to train live with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! It's not a secret that there is a growing demand for OpenSIPS Certified Professionals all over the world. This training is intended to offer you a 50/50 view between theoretical and practical applications of OpenSIPS. It covers advanced topics that are critical to telecom network operators in regards to security, scaling, and integration. For participants that would like an introduction to OpenSIPS basics and to qualify for prerequisites to the advanced topics, there will be a 2 day online primer prior to the live training.

    October 2015

    • 06 – 08 ITEXPO Anaheim 2015 - ITEXPO brings together service providers, enterprises, government agencies, resellers, vendors and developers to discuss the latest innovations that are changing the marketplace. Conference topics include WebRTC, cloud computing, unified communications, software defined networking (SDN), network functions virtualization (NFV), channel strategies and more. ITEXPO will be held October 6 - 8, 2015, at the Anaheim Convention Center in Anaheim, California.
    • 18 - 22 GITEX 2015 - 35th annual GITEX Technology Week in Dubai, UAE, 18-22 October. Speedflow will be among the recognizable exhibitors at the show, which acts as a hub for the Middle East, Africa and South Asia technologies industry.

    September 2015

    • 28 – 29 INDO AFRICA ICT EXPO 2015 - At Indo - Africa ICT Expo 2015, the latest innovative ICT technologies covering wide range of products, services, and software applications will be on display and open for technology transfer. ICT Expo scheduled at Nairobi, Kenya.
    • 16 – 18 Cloud Partners Conference - Cloud Partners Conference is a good platform to discover the cloud based solutions that can scale with your business needs. Cloud Partners Conference scheduled at Hynes Convention Center, Boston.
    • 01 – 04 Asian Carriers Conference 2015 (ACC) - The best and the biggest information technology and telecommunications companies from Asia and the rest of the world. ...

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  • 11/02/15--06:51: Mobile VoIP
  • Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

    How Does Mobile VoIP Work?


    Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

    By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

    Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

    A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

    Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

    • Anytime minutes
    • Night or weekend minutes
    • Rollover minutes
    • Roaming charges
    • Incoming call charges
    • Messaging limits
    • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

    Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

    Accessing Mobile VoIP

    Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

    Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. (The SIM card contains all the information needed to identify network subscribers. ...

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  • 11/02/15--08:28: Aastra
  • Aastra's SIP phones have multiline appearances and support functions like busy lamp fields and shared lines. They also have an XML API for application development and there are lots of free 3rd party applications available.

    Aastra also has a couple of small business appliances that are powered by Asterisk and Microsoft.

    SIP Phones

    The CT models include a cordless phone along with the base phone.

    • 67xxi Wallmount Bracket - Wall mount bracket which allows for simple and efficient installation of multiple models of Aastra IP phones.

    • 6735i - 5 line text display - 6 softkeys, 6 programmable keys, gigabit ethernet
    • 6737i - Dual 5 line text display, 12 softkeys, gigabit ethernet
    • 6739i - Color graphical display, resistive touch screen

    • 6757i - Dual 5 line text display, 12 softkeys
    • 6757i CT - Includes cordless handset (North America / Australia / New Zealand only)
    • 6755i - 5 line text display - 6 softkeys, 6 programmable keys
    • 6753i - 3 line text display - programmable buttons
    • 6751i - lobby phone

    • 6731i **New entry level phone
    • 6730i Similar to 6731i except no support for PoE and only 1 ethernet port.

    • 9480i - 5 line text display, 6 softkeys
    • 9480i CT - Includes cordless handset (North America / Australia / New Zealand only)
    • 9143i - 3 line text display, programmable keys


    Expansion Modules

    Up to 3 expansion modules are supported without extra external power supply.
    • M670i (previously 536M) - For the 53i, 55i, 57i, 57i CT and 39i. Provides 36 additional keys with LED status display. ...

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  • 11/02/15--20:13: SBO
  • sbo multipath logo.jpg

    What is SBO?

    Synchronous Bandwidth Optimizer is generally known as SBO. It is a bandwidth optimization solution for VoIP call termination developed by Synchronous ICT, a world famous VoIP software provider. It is a pioneer Bandwidth Optimization Technology fully loaded with amazing features.

    Why Bandwidth Saver is so important for VoIP?

    In VoIP termination, main expenditure is Bandwidth cost. Minimizing Bandwidth consumption is the main purpose of SBO Solution, at the same time it helps to reduce business operation cost. As a perfect bandwidth saver SBO reduces more than 80% internet cost directly. However, the quality of service is never compromised. To run a quality VoIP business congestion free, telco grade bandwidth connection is required . Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, SBO Multipath make its possible to run IP telephony business with low cost share internet connection by maintaining supreme service quality. It can combine multiple internet connections together and use all the available bandwidth creating a single connection.

    How does SBO work?

    Well, there are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes almost 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO works here. It has own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

    Key Features of SBO:

    • The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
    • SBO works behind any type of firewall and NAT. That means, it has anti-block feature.
    • Works with all commonly used codecs such as G.729, G.723 etc.
    • Multipath: SBO Multipath allows you to use multiple number of internet connections same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
    • Works with any type of internet connection i.e. GPRS, EDGE, 3G, 4G, Wi-Fi, Wi-Max, so it is possible to setup anywhere where mobile internet is available.
    • Highly secure and linux based software.
    • Full featured billing platform

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  • 11/02/15--20:38: Twitter VoIPUser Directory
  • This is a page that can contain entries of Twitter users and their business or VoIP interests.

    Please insert into correct alpha place and please leave things clean enough to be helpful to people on Twitter who want to find VoIP Users

    Voip-info.org Twitter: @Voip_info


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  • 11/02/15--21:55: voip-info.org
  • Welcome to the VOIP Wiki - a reference guide to all things VOIP.


    This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

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


    NEWS


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  • 11/02/15--21:59: New Software Releases
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  • 11/03/15--17:25: VOIP Phones
  • This page is for listing brief details of VoIP Phones including details, where to buy, specifications, and any other relevant VoIP Phone information. Please read the Posting Guidelines for Promoting Products and Services before adding to it.

    Hard Phones

    Standalone Ethernet Hard Phones (voice only)
    An Ethernet hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

    General



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  • 11/03/15--17:27: iKall
  • Please see iKall LAN Phones

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  • 11/03/15--17:28: VoIP Providers China
  • This page is a list of VoIP service providers in China. Please keep this list in alphabetical order. China VoIP providers looking to add their services can do so in the list below.

    • China SIP Trunking (CTS): A Leading Business VOIP Solutions Provider in China. Hosted PBX in China, SIP Trunking in China, DID Number, Toll Free Number, VOIP Consulting, VOIP System Maintenance, VOIP Architect, VOIP Specialist, Microsoft Lync Integration
    • Image
    • China Hosted PBX (CTS): A Leading Business VOIP Solutions Provider in China. Hosted PBX in China, SIP Trunking in China, DID Number, Toll Free Number, VOIP Consulting, VOIP System Maintenance, VOIP Architect, VOIP Specialist, Microsoft Lync Integration
    • BOL - VoIP Service Provider's Provider. Provides both H.323 and SIP softswitch, wholesale system, prepaid/postpaid realtime RADIUS billing, customizable softphone (encryption function included), softphone SDK, calling card, callback system and SIP Session Border Controller with encryption function. VoIP services: Wholesale(86), H323/SIP A-Z Route, Co-location at HongKong.
    • DIDWW:Chinese DID available now instantly on on-line store.
    • http://www.eztor.com-professional China VOIP provider.with high quality and low cost.
    • International Carrier Exchange, Room 701, Oversea Chinese, Venture Building, Shenzhen, China
    • Live-TechOffers TalkSwitch VoIP Hardware & Lynk VoIP Services | Remote Technical Support | VoIP Service Plans | Worldwide Support
    • Mydivert.com China DID Provider offers local VoIP DID Virtual Phone Numbers. A low cost virtual international telephone number with free incoming calls and call forwarding
    • http://www.vidanetwork.net/ip-pbx-bundle-en.php Reseller needed. Provides Business IP-PBX, ATA, IP-phone, DID numbers and service bundles in USA, Canada, China and India.
    • http://www.voipcallsecure.com VoIPCall Secure provides subscribers secure-encrypted phone service to businesses and high net worth individuals who want to keep their private phone calls private. Service available across the globe. Includes unlimited subscriber to subscriber calling and pay for service international calling.
    • http://www.voptech.comVoptech is a professional VoIP products provider,we have life long technical support,our products both competitive in price and quality.We also had been passed the Elastix compatible test already.IPLAN Argentine (http://iplan.com.ar/), a leading VoIP service provider in Latin America, orders several thousands of IP Phone VI2006 from us every month. We have VoIP Gateway with up to 96 FXS/FXO Ports. It had been certified by Mitel Canada (www.mitel.com), together with Cisco, AudioCodecs Gateways.

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  • 11/03/15--17:34: Asterisk sip canreinvite
  • Asterisk sip.conf, peer definition: canreinvite option

    Versions

    Migration from Asterisk 1.2 to 1.4: The "canreinvite" option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: "yes", "no", "nonat", "update". Please consult sip.conf.sample for detailed information.

    canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does. See also the closely related setting directrtpsetup.

    Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other.

    Description

    This peer option in sip.conf is used to tell the Asterisk server to not issue a reinvite to the client unless really necessary. This is used to interoperate with some (buggy) hardware that crashes if we reinvite, such as the common Cisco ATA 186.

    When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other.

    • If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
    • If the clients use different codecs, Asterisk will not issue a re-invite.
    • If the Dial() command contains ''t'', ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite.

    'canreinvite=no' stops the sending of the (re)INVITEs once the call is established. From messages in the archives and the Asterisk handbook one finds out that the Cisco ATA-186 does not handle the (re)INVITE well. This is necessary if the client and the Asterisk server is on opposite sides of a NAT gateway or firewall.

    • canreinvite = yes "allow RTP media direct"
    • canreinvite = no "deny re-invites"
    • canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
    • canreinvite = update "use UPDATE instead of INVITE"
    • canreinvite = update,nonat "use UPDATE when local, deny when NAT"

    Note: In spite of its name 'canreinvite' being set to 'no' does *NOT* disable all reINVITE operations. It *only* controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting!

    Notes

    • reinvite=yes/no is plain wrong, even if you see it mentioned in example .conf files. The correct syntax is canreinvite=yes/no
    • Connecting media paths direct to an endpoint behind NAT won't be pretty. Especially if both devices are behind NAT. You might want to try using SER's nathelper in conjunction since nathelper.so can rewrite the SDP so that the private IP addresses are not included in the re-invite.
    • When dtmfmode=rfc2833, asterisk will send the RTP stream through asterisk. With dtmfmode=info canreinvite works properly.
    • Asterisk 1.8 added the media_address= configuration option which can be used to explicitly specify the IP address to use in the SDP for media (audio, video, and text) streams.

    Background info

    In general Asterisk supports 3 methods of media handling:

    1. ...

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    Ring CO3 on Pansonic KSU for VoicePulse


    • From iax.conf

    [voicepulse]
    context=voicepulse-in
    username=loginname
    secret=loginpasswd
    auth=md5
    type=friend
    disallow=all
    allow=gsm
    allow=ulaw
    allow=alaw
    allow=ilbc
    host=gw5.voicepulse.com
    nat=yes

    • From extensions.conf

    [voicepulse-in]
    ;
    ;
    ;
    ; This routine handles inbound calls on our VoicePulse DID
    ; Voicepulse rings in with theextension set to our DID
    ;
    exten => _7035556709,1,SetGlobalVar(RINGINGLINE=custom/on-line-three)
    exten => _7035556709,2,Dial(ZAP/1,${DEFTIMEOUT},Tt)
    exten => _7035556709,3,Goto(office-menu,s,1)
    exten => _7035556709,103,Goto(office-menu,s,1)
    ;
    ;


    Ring CO4 on Pansonic KSU for BroadVoice


    • From sip.conf

    ;
    ; register with BroadVoice VoIP service
    ;
    context=broadvoice-in
    externip=ghend-pbx.homeip.net
    dtmfmode=inband
    register=>7035551557:mypasswd@sip.broadvoice.com
    qualify=yes
    ;
    [broadvoice]
    type=friend
    username=7035551557
    fromuser=7035551557
    secret=mypasswd
    host=147.135.8.129
    context=broadvoice-in
    fromdomain=sip.broadvoice.com
    srvlookup=yes
    insecure=yes
    nat=yes
    dtmfmode=inband
    canreinvite=no
    disallow=all ; First disallow all codecs
    allow=ulaw ; Allow codecs in order of preference
    allow=gsm
    allow=alaw
    allow=ilbc ; Note: codec order is respected only in [general]
    ;
    qualify=yes
    ;
    ;

    • From extensions.conf

    [broadvoice-in]
    ;
    ; BroadVoice rings in on the sextension
    ;
    exten => s,1,SetGlobalVar(RINGINGLINE=custom/on-line-four)
    exten => s,2,Dial(ZAP/2,${DEFTIMEOUT},Tt)
    exten => s,3,Goto(office-menu,s,1)
    exten => s,103,Goto(office-menu,s,1)
    ;
    ;


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    ConfBridge

    Synopsis

    ConfBridge conferencing bridge

    Description

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

    10.x

    ConfBridge 在新版的 Asterisk 10.x 做了許多更動。新版的文件可見於此 here. (Someone should add it to Voip-Info)

    Asterisk cmd ConfBridge


    ConfBridge此應用程式於 Asterisk 1.6.2.* 開始就有了, ConfBridge 功能上與 MeetMe 相似,但不同於 MeetMe的地方在於 ConfBridge 不需使用 DAHDI 硬體混音。 取而代之的是,混音直接在 Asterisk 內部直接做了。

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

    參數:options可以為空,也可以設置一個或者多個參數,參數有如下一些:
    • 'a' — 設為管理員
    • 'A' — 設為標記成員
    • 'c' — 計算有多少匿名成員進入會議室
    • 'm' — 設定一開始就為靜音模式
    • 'M' — 啟用音樂等候,當會議室只有一人時,可特別指定音樂等候的等級,若未指定則使用目前頻道的設定或預設值
    • '1' — 不撥放提示訊息,當第1位使用者進入時
    • 's' — 當收到 # 號鍵時,出現語音選單 (普通成員是普通選單、管理員是管理選單)
    • 'w' — 等待標記成員進入後會議才開始(若有啟用 M 參數,則在那之前都會一直撥放等候音樂)
    • 'q' — 靜音模式 (不撥放進入/離開訊息)

    進入聲音可用 'CONFBRIDGE_JOIN_SOUND' 此變數,
    離開聲音可用 'CONFBRIDGE_LEAVE_SOUND' 此變數
    適用於個別使用者,可為他們配置不同的音檔

    註: 此應用程式不會自動應答該頻道

    Muteing

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

    不同於 MeetMe

    不能再象這樣子 MeetMe(123,d,321) 的設定會議室密碼,取而代之的是,可在外部完成此一設定,因為很可能在兩人同時輸入密碼時程式當掉, it is not possible to use the CONFBRIDGE_INFO(parties) function and the code must, at least in part, be run in a MacroExclusive() application. ...

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  • 11/03/15--17:40: Asterisk Consultants Greece
  • Asterisk consultants: Greece


    Modulus SA

    VoIP Service Provider
    Virtual PBX, SIP Trunking, Greek DID numbers, VoIP Termination, Custom application development that requires VoIP technology, Asterisk PBX


    Netvoice.gr

    VOIP solutions & IT Services

    VOIPsystems.gr

    Advanced VOIP telecommunication solutions based on Asterisk.

    MerIT Advanced IT Solutions, Athens

    Voip Telephony, Servers & Corporate Networks, High Performance Computing.

    Transcom Ltd., Athens

    Voip Telephony, Asterisk Intergrators

    SEQUENCE ,Athens


    Application Service Provider, Telephony Software development, System Integrators, Networking Solutions, Asterisk Custom Applications.
    Products:QUATTRO Provider -Contact Center- Corporate | CONCEPT Ivr - UM -Sip Proxy | Asterisk CTI QUATTRO |SMS INTERACTIVE GATEWAY

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  • 11/03/15--18:09: Business IP Phone System
  • CooVox Series IP Phone System

    U-2.jpg

    CooVox-U20: Smart GSM IP PBX


    • No license fee
    • GSM or 3G optional
    • Monitor and whisper
    • 10 concurrent calls
    • 30 IP Phone registrations/ extensions
    • Many advanced IP PBX features
    • Virtual Fax, Voicemail to Email
    • Call Detail Records(CDR)
    • Automatic Call Distribution(ACD)
    • Enhanced security policies
    • H.D. Voice(G.722) supported
    See more details: http://zycoo.com/html/U20.html

    CooVox-U50: Hybrid IP PBX

    • No license fee
    • 3G optional
    • 12 concurrent calls
    • 100 IP Phone registrations/ extensions
    • Virtual Fax, Voicemail to Email
    • Call Detail Records(CDR)
    • Automatic Call Distribution(ACD)
    • Enhanced security policies
    • H.D. Voice(G.722) supported
    • Modular design_Plug and play
    See more details: http://zycoo.com/html/U50.html

    CooVox-U100: Enterprise IP PBX

    • No license fee
    • 3G optional
    • 80 concurrent calls
    • 500 IP Phone registrations/ extensions
    • Firewall
    • Remote backup service
    • Instrusion detection
    • Virtual Fax, Voicemail to Email
    • Call Detail Records(CDR)
    • Automatic Call Distribution(ACD)
    See more details: http://zycoo.com/html/U100.html



    http://www.zycoo.com
    H.Q.: 7F, B7, Tianfu Software Park, Chengdu, China.610041
    Tel: +86 28 85337096 ext.806
    Fax: +86 28 85337096 ext.800
    Email: zycoo@zycoo.com


    Zycoo Argentina

    VoIP Experts S.R.L. - Distribuidor Exclusivo Zycoo

    Te: +54 (11) 52190914 +54 (351) 5682878
    skype: voipexperts
    www.voipexperts.com.ar / info@voipexperts.com.ar





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  • 11/03/15--18:14: IVRLuke
  • Recording prompts


    For your IVR configuration you will need first to record some sound messages.
    You can record your own sound files using the Asterisk. For this purpose you should add an extension for the recording in the extensions.conf file. In our example, you can dial the extension 100, after the beep you can start recording your message.
    To end the recording press "#". The message is recorded in .gsm format and is called recording.gsm in the /var/lib/asterisk/sounds directory. Asterisk plays back the message after you pressed "#"(after 2 seconds).

    
    exten => 100,1,Wait(2)
    exten => 100,n,Record(/var/lib/asterisk/sounds/recording:gsm)
    exten => 100,n,NoOp(${RECORDED_FILE})
    exten => 100,n,Wait(2)
    exten => 100,n,Playback(/var/lib/asterisk/sounds/recording)
    exten => 100,n,NoOp(${PLAYBACKSTATUS})
    exten => 100,n,Wait(1)
    exten => 100,n,Hangup()




    Here we are using Wait(), Record(), Playback() and Hangup() applications. You can see their descriptions by typing "core show application <application>" in Asterisk 1.4, or use "show application <application> (this will be removed in future).

    Let's make more clearness:
    Wait(2) - waits 2 seconds, then the dialplan continues to the next priority. The 2 seconds are to be sure that the channel is up and ready for recording.

    Record(file:gsm) - records from the channel into a file with format gsm. This application sets a channel variable named RECORDED_FILE, which contains the name of recorded filename.
    You can use NoOp() application to see what contains the RECORDED_FILE variable.

    exten => s,n,NoOp(${RECORDED_FILE})

    Playback() application - plays the recording file. This application sets the following channel variable - PLAYBACKSTATUS. The result is SUCCESS or FAILED.

    Hangup() application - hang up the calling channel.



    Implementing a IVR using menu structure


    The following is an example for an IVR system. You dial any number in the range [1-9] and you enter the IVR system. First there is checking if it is business hours and if you are, you will be send in the ivr-lang context, after that you will get the greeting message and during the message you can press a key from one to three to choose your language. According the number you pressed, you will be directed to the right extension (context).

    
    [incoming]
    
    exten => _0[1-9].,1,GotoIfTime(9:00-18:00|mon-fri|*|*?ivr-lang,s,1)
    exten => _0[1-9].,n,GotoIfTime(10:00-17:00|sat|*|*?ivr-lang,s,1)
    exten => _0[1-9].,n,Playback(closed)
    exten => _0[1-9].,n,Playback(closed)
    exten => _0[1-9]. ...

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  • 11/03/15--19:34: Voice Broadcasting
  • Voice Broadcasting

    The following is the definition for Voice Broadcasting:

    • "Voice broadcasting is a mass communication technique, begun in the 1990s, that broadcasts telephone messages to hundreds or thousands of call recipients at once. This technology has both commercial and community applications. Voice broadcast users can contact targets (whether they be members, subscribers, constituents, employees, or customers) almost immediately." - Wikipedia

    Voice broadcasting phone software manages a database of phone lists as well as digitized phone messages. Using analog, digital or VOIP telephony components, these computers can simultaneously broadcast thousands of phone messages. Personalized information can be included in the phone messages through the integration of text to voice software.

    Advanced systems include telephony boards or software that can detect the difference between an answering machine and a 'live' person answering the call. These systems employ the logic to properly play a unique message to answering machines without message truncation.

    There are few enterprise-level commercial solutions for Voice Broadcasting. They include such functions as:
    • Possibility to play personalized messages to every recipient which consist from library of predefined messages (pre recorder by every client) and some variable information (dates, numbers, digits, amounts with currency and so on). This method eliminates discomfort from text to voice modules (message looks completely like pre recorded by a human) and have high flexibility within predefined applications.
    • List of phone-numbers to call with parameters of call in DB such as Oracle DB, Microsoft SQL, IBM DB/2, MySQL, PostgreSQL.
    • Possibility to organize polls and to combine poll with statical or dynamical message. All results of such polls store within enterprise grade DB.
    • Possibility to transfer to call center by DTMF command from called party.
    • Possibility to opt-out by DTMF from called party.
    Such solutions always customizable for every client and easily integrates with ERP, CRM, billing systems. They usually used by banks, collectors and credit companies in order to remember their clients about payment day (with personalized amount to pay calculated to specific day).

    VoiceXML and Voice Broadcasting

    Phone messaging systems and services automatically send phone messages from a computer system to a remote phone systems using XML push logic. For example, alerts can be broadcast to tenants in a building if there is a fire or disaster. Heat sensors with IP connectivity that are installed in the tenant building can initiate an emergency voice broadcast by sending a VoiceXML message . Other applications may be as simple as wakeup calls or weather alerts that are triggered automatically from a computer system or websites.

    VoiceXML Gateway Information

    VoiceXML is a W3C standard for interactive telephone applications. The platform/server for VoiceXML is called a VoiceXML gateway. The gateway is like a web server, but it interprets VoiceXML command instead of HTML. You can test out VoiceXML applications with a hosted VoiceXML service, you could also setup your own VoiceXML gateway.





    See Also (Vendor Information)


    Voice Broadcast Applications


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  • 11/04/15--06:10: SIP Trunk Providers India
  • This page is a list of SIP trunk providers in India. Please keep this list in alphabetical order. India SIP providers looking to add their services can do so in the list below.

    • ALTOTELECOM - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com

    • Incorpus TeleNetworks Your one stop solution to all voip needs, Call termination, resellers, DIDs, Softswitches, Tollfree, residentail voip, Call center voip.
    Contact our executive today for more details visit our site and have a live chat or mail us at info@incorpus.in

    • AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.

    • CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com. Contact Number: 800-231-9802

    • i7 Solutions - VOIP Wholesale and Retail i7 Solutions is looking for VoIP Resellers in markets all around the world. We provide the Dialers and Rates your customers want, and the tools for you to control all aspects of your business. i7 Solutions is a one of the very few global VoIP solution providers with direct contracts with local incumbent telecommunications operations across the globe. Our primary business includes A2Z termination services, retails & reseller solutions for VoIP services and also providing calling cards. India TATA CLI is our premium route. Please contact us @ Gtalk and Email ID info@i7solutions.in

    • UtterU UtterU is leading online voice calling business service provider. UtterU provide Wholesale option for reseller to start international calling business around the world. UtterU also provide SIP trunk facility for all business to making call over the internet at very low cost.

    • http://www.doorVaani.com DoorVaani.com is a Business, Residential and Personal VOIP services provider offering VOIP Call Minutes to destinations worldwide, Local Phone Numbers or DIDs in about 60 countries and Toll-free numbers in 5 countries. DoorVaani.com is also an automated web application where ordering of services, making payments and provisioning is done all in real time with no wait time or human interaction needed. Online payments are accepted in 25 currencies including Indian Rupees (INR). Offline payments or Bank Transfers accepted in INR and USD.



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  • 11/04/15--10:29: Voicepulse
  • VoicePulse


    Introducing VoicePulse for Business.

    Company News

    2015-11-04 Blog Post: How to choose a business phone system
    2015-08-11 VoicePulse launches CallCode, a call flow feature that allows users to write Javascript code to integrate with third party platforms.
    2015-07-17 New Feature Release - Real-time 24 hour usage, Usage Limit Reset, Calling Access
    2014-11-06 VoicePulse Introduces VoicePulse FIVE, the Next generation in VoIP Services


    VoicePulse for Business

    Our hosted PBX solution runs in the cloud so you don't need an on-site PBX to provide voicemail, call recording or other features. Reduce costs and free up your IT resources to work on things that really matter.

    Unlimited Calling seat as low as $24.95 per month (based on a 2 seat minimum.) Per minute pricing available, please contact sales.

    Features such as
    Unlimited US Calling, including DID, Voicemail, E911 and Toll Free DID
    Conference Bridge
    Hold Music
    T.38 Fax Line
    Hosted Fax3
    CallerID Routing
    Ring Groups
    SLA Groups
    Time Frames
    Holidays
    Listen / Whisper
    Dial By Name Directory
    Intercom
    Directed Pickup
    Call Parking
    Auto Attendant
    Call Recording4
    Call Center5
    Plugins6
    DISA and more

    Buy, rent, or bring your own feature-packed IP phones from manufacturers like Polycom, Cisco, or Yealink.

    Watch our video about how a cloud PBX can grow your business

    VoicePulse SIP Trunking

    Business Gateway Pricing click here.

    Ideal for SMB and Enterprise

    Use your existing SIP enabled on-premise softswitch or PBX to make and receive phone calls regardless of call volume. Choose your number from anywhere in the US or move your existing DIDs to VoicePulse FIVE. There is no fee to port your phone number to VoicePulse FIVE.

    Rates

    Incoming calls to a U.S. phone number are $.01 per minute. Outgoing calls to the US48 are $.02 per minute. Incoming calls to a U.S. toll-free phone number are $.029 per minute. Look up
    international termination rates here.

    Account Center


    • View your current Statement Balance
    • Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
    • Make instant payments by credit card
    • Add unlimited Channels or call paths to your Trunk
    • View your active phone numbers
    • See our inventory of numbers
    • Instantly activate new numbers
    • Manage E911

    Supported Protocols


    • Session Initiation Protocol (SIP)

    Supported Codecs


    • G.729a
    • G.711ulaw
    • G. ...

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  • 11/04/15--12:46: Xorcom
  • Xorcom Ltd.

    Xorcom+slogan white background.bmp

    Manufacturer of high-quality Asterisk hardware since 2004:

    Our Product Lines:

    • Astribank - The first channel bank designed for Asterisk.
    astribank-angled.png

    Features:
      • Up to 8/16/24/32 FXS/FXO ports per 19" 1U wall- or rack-mountable unit
      • Up to 8 BRI ISDN ports
      • Up to 4 E1/T1 (PRI, R2, CAS) ports
      • Connects via USB 2.0, no need for additional PRI card, USB 2.0 hub may be used
      • Features input and output relays for connecting and operating external devices
      • Easy to install and implement


    xt3000-angled-600.png

    Features:
      • Various combinations of FXS, FXO, BRI and E1/T1 (PRI/R2/CAS) ports
      • Supports SIP and IAX2 phones and trunks
      • From 8 to 32 analog lines/extensions integrated into the unit
      • Up to 4 ISDN E1/T1 (PRI, R2, CAS) ports
      • Two hot swappable disk drives
      • Two built-in redundant power supplies
      • Support for up to four Astribank USB channel bank units internally (24 Astribanks total)
      • Up to 480 concurrent calls (550 if SIP-only)
      • Dual hard drive (RAID1)
      • Redundant fans for cooling
      • Internal backup and restore
      • Front panel USB access
      • Two Ethernet ports to allow separation of IP voice and data traffic for improved voice quality and increased throughput
      • Supports auxiliary appliances (door locks, alarm systems). Available for models that feature I/O ports.
      • FreePBX™ - Easy-to-use Web interface for Asterisk and network setup
      • Advanced support and maintenance features:
        • A multi-function LCD (Liquid Crystal Display) to perform the most common functions directly on the front panel of the IP-PBX without having to attach a keyboard and monitor.
        • Rapid Tunneling™ - Provides direct support for customers via a secure connection, behind firewalls and NATs
        • Internet updates
        • Configuration export / import

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