Voice over Internet Protocol or VoIP refers to a host of transmission technologies meant to deliver voice communications over IP networks that include packet switched networks and Internet. VoIP systems include communication services such as facsimile, voice, and voice messaging applications. A number of firms are shifting to this system to promote a cost effective medium of communication. You can hire an expert who can help your firm to leverage the potential of this new system. Just post a job today! VoIP Developers 분야 채용

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    Trophy icon Asterisk Call File With DTMF 2 일 left

    Hello / Good evening everyone, I would like you to help me create an asterisk call file which places several dtmf commands in the call file channel once the call is connected. The DTMF does not necessarily have to go through the call file. If you want you can use the [로그인하시면, URL을 확인하실 수 있습니다.] file, but my goal is to place dtmf commands in an ongoing call generated by a call file.

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    Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile ver...

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    Full Asterisk IVR Setup training and cdr report with below requirement 1. Creation of IVR flow chart and diagram 2. Communicating with other stakeholders for the Audio files. 3. IVR Configuration on FreePBX 4. Setting up Inbound routes to point to IVR (via GUI and Command line) 5. Other tweaks on IVR (schedules / time conditions) (via GUI and Command line) 6. Setup of dial plan on GUI and CLI 7. ...

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    Kazoo PBX development 6 일 left
    확인

    We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of telephony systems based on these platforms -Excellent understanding of pbx telephony, dial-plans,...

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    If you can create an add-in in outlook and understand SIP/VOIP please reach out.

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    Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. Thank you.

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    COLD CALLING -- 2 5 일 left
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    We need help with cold calling businesses within the United States to help locate sellers of specific equipment, and compile a list of the results.

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    Hi, I need to set up a rule that in FreePBX that will allow changing the CID of the extension based on the outbound route. For example, if I dial out from exten 200, through outbound route 1 it will get CID1, outbound route 2 - CID2, etc. The outbound routes are NOT defining the CID, the extension defines the CID.

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    Sip Client 5 일 left
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    Device Compatibility: The App will be compatible for Android devices. Scope of work : ● revamp the open source code and redo the entire application ● Remove/ disable extra functions ● Revamp the design ● Update new color combinations and logos ● Test the application ● Deploying It to store SDK linphone

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    I am in need of resources and tutorial on kamailio configuration file for serial forking with tutorial

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    Astpp billing setup 5 일 left
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    I have installed Astpp on centos 7.x server and need to confiugre it with fusionpbx that is installed on separate server , you need to perform following make setup functional and test create user / tenant / domain add sip trunks create extensions successful testing cooperation rebranding etc

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    Hello! I'd like to invite you to take a look at the job I've posted. Please submit a proposal if you're available and interested. Telecom specialist We are looking for system integrator who can help us to integrate facebook magma open source EPC with Mikroitk Enodeb .Its 4G LTE Private network project. We have one test unit from baicell which is recomanded by facebook . Our ...

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    i need to stop fake ringing on linphone application i used sip account to make a call

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    I am having problems with extension loosing register, i dont have a very good internet connection but i am almost certian that i can fix this problem, customer has 50mbps upload and download, the problem is the pings not being very stable, due to a wireless ptp connection, freepbx / asterisk setup.

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    I have sms working with flowroute and my fusionpbx. But right now its only working in one domain. I would like to make it work will all my domains and each domain with its own flowroute account. I have test server prepared for the task already.

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    I am looking for a telecom expert, especially familiar with ZTE NGN Core equipment who will help us re-dimension/re-purpose our SoftSwitch-es decreasing the resources dedicated for the TDM and reallocating those resources for VoIP. the model and version of the soft-switches is ZXSS10 SS1b [로그인하시면, URL을 확인하실 수 있습니다.] it is important to have zero downtime, as the device is in production, and also to...

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    Program needs to parse large lists of phone numbers in an Excel Spreadsheet to determine which are currently "active lines". Once an active line is found, the program needs to be able to either: a) do a "speech-to-text" conversion to copy the voicemail personal greeting, or b) record the audio of the voicemail greeting and have it identified as being connected to a certain ph...

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    I need a prototype functional application from a mobile phone which captures audio and send it back to a server (LINUX or WINDOWS) where it gets mixed with other audio from other live cellphones and create a single audio source to stream. It needs to receive the feedback from around 2 to 5 users at the first phase This will be used to create a chorus concert online. It needs to be a functional...

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    I need call recording method, which records VOIP call, which is integrated in my android application

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    Hi, We have a PRI line and connected to SIP server using FreePBX. Currently our employees working from home and using some soft phones but there is no voice clarity. We would like to setup a virtual callcenter which redirect to their mobile numbers. If this is possible please reach me.

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    Need someone available in KSA Riyadh, We need to configure the Yeaster Part .

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    PBX Business Phone System 2 일 left
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    Looking for business pbx phone system. Cloud based here is a reference website [로그인하시면, URL을 확인하실 수 있습니다.] Mobile App Desktop App call waiting conference calling call transfer auto attendent extensions etc

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    We are a well established company in the hotel industry in USA for Telecom System, Network and CCTV installations and ISP services. We are looking for seasoned individuals for Base pay and commissions for telemarketing our products and generating leads. We will be giving you list to call if needed.

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    Need help calling competitor companies to determine wholesale pricing

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    Hello! Looking for someone to write an app to auto dial into a conference call. I can give more details once we move forward. And dont ask what language I want it programmed in, If I knew that I would do it myself :) You tell me!

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    Hello, I need someone to help us by setting up an inbound call center using Bitrix24 dialer. We have our own sip voip trunk and can also host freepbx vps if needed. We need to achieve the following call flow: Customer calls —> pbx places call on hold while waiting for InGroup member to be available —> once InGroup member is available call is routed to them in the Bitrix24 diale...

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    IPRN BUSINESS 2 일 left
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    looking for tech person who has previously worked in IPRN (iprs) business international premium rate numbers

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    We have a lot of gateways for SMS services. All of them acept AT commands but, only a few have a VOIP access. (SIP protocol). For that, we need an app to convert SIP protocol to AT commands for termination of VOIP calls over SIMs through AT commands.

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    hello i am looking for someone with freeswitch knowledge person with astpp (billing) will be very good. i need something to do with SIP TLS with letsencrypt and other voip work.. so you need be very good on freeswitch for this

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    We're looking for a network administrator who has experience with core network infrastructure, Cisco, Juniper, Mikrotik, Freeradius, Nokia, Alcatel Lucent. Administrator will be responsible to create new core network infrastructure as required, preconfigure Nokia / Alcatel GPON networks.

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    Asterisk, Issabel 1 일 left
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    Hacer modificaciones a la plataforma de Issabel: 1.- Poder identificar quien corta la llamada, cliente o agente 2.- Marcar hasta 4 números por registro 3.- Mensaje emergente cuando el agente recibe una llamada de una cola especifica 4.- Hasta que el agente guarde datos le caiga la siguiente llamada 5.- Bajar el tiempo de revisión de llamadas agendadas 6.- Poder tener hasta 3 llamadas...

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    Looking for Asterisk expert who can help configure PBX. This task probably needs LESS than an hour work in this case my budget also mirroring. You need to able to talk.

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    Need to install configure and test [로그인하시면, URL을 확인하실 수 있습니다.] on Debian.

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    We need a solution to replace our FreePBX setup. We currently have 4 instance of FreePBX, but this is not the perfect solution. Instead we would like two Class 4 softswitch for redundancy. in the front to give LCR and SBC fonction. We figured that Kamailio is one of or best solution and would then add Class5 switch, We would think that FusionPBX is a better solution than FreePBX. The setup need to...

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    I am looking for a fail over to 4/5G when the primary service (Ethernet, XDSL, NBN) fails. This also needs to transfer the SIP calls to the 4/5G backup service. Currently there are a lot of backup services available but i have not come across one where the calls are transferred across. o do this the static IP address will need to be moved to 4/5G service.

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    help with Twilio.com 18 시간 left

    Help setting up a voice api on [로그인하시면, URL을 확인하실 수 있습니다.]

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    Need to install sems-server on Debian server side and Openwrt client side. UAsL <---SIP/RTP---> SEMS-SBC1 <===SIP/MUX RTP===> SEMS-SBC2 <---> UAsR SEMS-SBC2 will be behind NAT. [로그인하시면, URL을 확인하실 수 있습니다.] [로그인하시면, URL을 확인하실 수 있습니다.]

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    Create IVR System 15 시간 left
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    Create IVR System for my Mobile,

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    as title need asterisk guru support for a call center project and little small distribution

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    O sistema do Issabel ele lê do o nome do cliente se tiver em uma lista? Ele ouve a resposta do cliente como sim ou não e me computa isso?? Preciso que leia o nome do cliente e depois fale o valor que o cliente tem disponível e depois ouça a resposta sim ou não é me dê um relatório. Tenho uma empresa de empréstimo consignado

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    Whatsapp gateway 6 시간 left

    Hello, Looking for SIP -> Whatsapp call gateway, it should take the calls from SIP (Freeswitch/Asterisk) and terminate to whatsapp number. Status of number should also monitored on whatsapp that means, if number is active on whatsapp, furthermore if it's getting the rings or not. It should also return the correct call error codes CALL SUCCESS, BUSY, UNAVAILABLE, etc.

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    About Telp Telp has two businesses one that operates a small call centre with 2 to 6 casuals and another which focuses on being a partner or reseller of Zoho suite of solutions. About the role We require someone who has a demonstrated technical background, and that is proficient in IT systems and solutions, and ideally has experience in setting up VoIP, softphones, telecommunication systems, and ...

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    I want a web application with android and desktop interfaces for conducting live lectures for Teachers - Students: (example: unacademy/edx / etc) 1. To have one to one, one to many , many to many video conferencing, 2. White board functionality with marking and other functions. 3. File transfer, raise hand, send question, answer, features 4. Ability to see old lectures. Cloud can be used. I w...

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    VOIP Freepbx 종료 left
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    I am a system integrator and manage systems for my customers . I am setting up a VOIP system for one of my customers using freePBX and would like to get assistance in setting it up. I am familiar with freePBX and how to use it, I need someone who can assist me with troubleshooting and provide additional support when required. Ideally this would suite someone already in the VOIP area and currentl...

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    i have 3 x Skype accounts running daily on windows 7 ...i require one of the accounts to run on my Mac OS 10.8.5 without effecting or losing messages etc ....you will need Teamviewer URGENT

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    Skype Expert 2 일 left
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    need to run 3 x simultaneous Skype accounts on windows 7 with 3 dedicated skype numbers and keep all contacts and messages .......is a mess! I know :-)

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    VoIP 커뮤니티 게시글 (최상위)