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    13,044 asterisk pbx server công việc được tìm thấy, giá USD

    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Soft Switch development. Đã kết thúc left

    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Soft Switch development Đã kết thúc left

    Asterisk based Soft Switch [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Ajuste e atualização do site Đã kết thúc left

    Atualização do site, segue as atividades| 0.1. Conversor de tabelas para Rocket (central asterisk) | 0.2 Integração entre o Rocket (central asterisk) e PROSPECTS (area administrativa) | 0.3 Inclusao do Te Ligamos (click to call) e botao blog no menu area cliente do site 1. Blog (do 0) | 2. Integração com Rd Station | [đăng nhập để xem URL] de...

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    vicidial message upgrade Đã kết thúc left

    ...I have 110,000 phone numbers to call bi-weekly. The problem I need to overcome is detection as to whether I am calling: 1. a basic residential type answering machine or 2. a PBX style answering machine which has 4 different ways of asking you to leave a message by pressing 0 or leaving a message for extension 101 or other combinations. The problem with

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    voice mail program Đã kết thúc left

    ...have 110,000 phone numbers to call bi-weekly. The problem I need to overcome is detection as to whether I am calling: 1. a basic residential type answering machine or 2. a PBX style answering machine which has 4 different ways of asking you to leave a message by pressing 0 or leaving a message for extension 101 or other combinations. The problem with

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    webrtc asterisk Đã kết thúc left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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    ...configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with other freeswitch, asterisk and fusionPBX installations. The task is to look at my setup, determine the problem, provide the solution and provide documentation of the problem and solution. Note 1. You

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    ...integrate Asterisk server with incoming and outgoing call center for our business in US. 1. Need a person who knows exactly how to do that using asterisk and what other external integration options we will needed to integrate. 2. Hardware requirement for implementation. 3. We should be able to make call from API over webrtc using asterisk underlying

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    ...client and admin portal. Users can register and fill in personal details. It connects to a MsSQL db There are links to a SMS gateway, as well as integrations with ASTERISK phone server and Stripe payment gateway The application is similar to an accounting suite, where invoices and other payments are entered. The application then runs a payroll process

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    I need a cloud pbx program for telephone. it needs to have a menu and some sub menues, take numbers as imput and store them in a database. record audio files, play audio files, have text to speech that will play back based on rules. take caller id, and store all the information in a database. i am uploading a sample file with the program that i wrote

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    I need SMS Gateway Đã kết thúc left

    Hi, I need SMS gateway, Without using any third party services like twilio etc. I need my own gateway. Freelancer must have experience with Voip/PBX/SIP etc

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    Need someone to install, configure and test asterisk a2billing (for calling card service use) on a Google Cloud VM. Only bidders with proof of prior/similar work will be considered. Asterisk realtime experience a plus. Thanks.

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    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

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    I am looking to create a SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own)

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    I have A Server and server have 2 Real IP IS ON USA i have a openwrt Router its Bangladesh and router have connected a VOIP DEVICE i want to send call THIS VOIP DEVICE now i have 2options 1) Install VPN on server and openwrt ROUTER and bridge network and set a static IP same serial on VOIP DEVICE useing MASQUERADE 2) make remote connections and USE

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    We are looking to skillful guys to Configure 3 SPA3000 devices & 1 GSM gateway with xCALLY contact cloud center (Asterisk base call center system). The job will be mainly sitting the devices to be connected with xCALLY systems using the sitting parameter of xCALLY GUI. * the xCALLY is hosted in VPS @ germany. * The VPS @ germany is connected to NordVPN

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    vos3000 expert Đã kết thúc left

    I need an expert to give me some tutorials on how to configure my vos3000 with my buyers and sellers details along with the rates and how to secure my vos from being hack...on a daily part time basis at $11 AUD an hour. This part time employment will be on a long term basis. After your tutorials in vos3000 i would also like some tutorials in Asterisk.

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    chan_dongle web interface Đã kết thúc left

    i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not

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    pfsense expert -- 4 Đã kết thúc left

    ...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud

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    ...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message

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    support Asterisk Đã kết thúc left

    need new server bassed on asterix

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    Project for Francisco M. Đã kết thúc left

    asterisk click to call web portal

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    a2billing / asterisk experts Đã kết thúc left

    Hi, we are in need of a asterisk / a2billing developer to help out on a small fix . please message to find out the details.

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    ...sent to you has the new technologies we are looking for, Back-end IVR integrations with Avaya contact center PBX -server fully blended Multi-channel support for voice, Email, im, Web chat, SMS, and social media contact types( Telephony server System Avaya) integration with the new Car Rental, Chauffeur and Coaches system database. The new Car Rental

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    I need to configure Vtiger Asterisk Connector to do click to call (outbound and inbound calls). I have my Vtiger installed and my own Pbx

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    добрый день! Нужен специалист по докеру. Есть задача по тиражированию сервера Asterisk. Нужно быстро копировать/сохранять копию/ дублировать Asterisk. Давайте выйдем в скайп, я подключу нашего технического директора и подробнее обсудим задачу. [Removed by Freelancer.com Admin]

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    Content writing for my website. Đã kết thúc left

    ...write content that's suitable to use at website at [đăng nhập để xem URL], these are the pages: 1 home page, bottom part 6 pages (1 main + 5 sub pages) under [đăng nhập để xem URL] 4 pages (1 main + 3 sub pages) under [đăng nhập để xem URL] 3 pages (1 main + 3 sub pages) under [đăng nhập để xem URL] 1 page under [đăng nhập để xem URL]

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    I need my gateway to be set up so that I can receive international voip traffic and forwarded using the simcards on the gateway. You set the route and Asterisk server on VPS. This should take just few minutes

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    I am looking to hire someone to link Asterisk with AWS Lex or [đăng nhập để xem URL] and add a Twilio or similar provider. I need an easy interface to update Lex or API.AI. This will include the ability to edit complex dialog flow through a user interface. The ability to route different customer phone numbers to and from different AI flows. When the AI flow does

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    Need a browser extension or application for both windows and mac. The soft phone will require an agent to log in and be able to show when an agent is ...agent to another agent. All call details will need to captured - Date & Time, Caller ID (Number) Start of Call and Finish, Agent ID Needs to be able to work across SIP and PBX providers using TAPI

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    Multitenant Asterisk PBX Đã kết thúc left

    Asterisk development with HTML5 interface. - admin web interface to create corporations/tenant PBX - customer web interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation

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    [đăng nhập để xem URL]

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    PBX personnel Wanted Đã kết thúc left

    Please apply with my post, we need someone who can help us with PBX issues.

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    Hell...single field from out PHP program to a linux/asterisk script/process that will: 1. Take the number as an input value 2. Attempt to call that number provided for 1 second from Asterisk/Trunk 3. Catch the return code of the attempted call 4. Pass the return code back to the calling PHP program We are using Asterisk 13.19.1 and FreePBX [đăng nhập để xem URL]

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    ...calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via the SIM installed In a way or another this APP will make the mobile work

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    A2billing with asterisk installation Script + Configure the basic calling card and DID function

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    Configure SPA3000 with asterisk Đã kết thúc left

    i Need to configure FXO/FXS gw (SPA3000) with asterisk. the interface system is xCALLY. i have connected SPA3000 with my DNS (fixed domain from [đăng nhập để xem URL]) the connection between asterisk server and SPA3000 DNS/IP should be via VPN which already setup on asterisk server i have uploaded 2 image showing the xCALLY trunk steps/inputs.

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    SIPSTATION FREEPBX no audio. Đã kết thúc left

    i recently got my number ported to sipstation and we have no audio both ways. I opened all the correct ports and also tested on a hosted pbx system. still no audio. I am lost and need help

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    ...sure i can use a SIP account as a routing gateway I want to make same to same VOS3000 softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work you can use any good Engine like asterisk,FreeSWITCH,Kamailio,OpenSIPS,etc More info check Attach file here is all screen shoot i need a complete wholesale

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    I have an asterisk server: Asterisk 1.8.20.0 built by palosanto @ [đăng nhập để xem URL] I need a php based script which will popup a browser window for every incoming call & show information based on QUEUE & DNID. Budget: $50

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    SIP failover using SRV records Đã kết thúc left

    ... Need somebody to provide me a proper configuration tutorial and example of usage. My scenario is: Have two SIP servers in two different locations. Both running Vodia SIP PBX. Vodia using multi tenant subdomain structure based on sub-domains. Each tenant is a sub-domain. DNS provider is Godaddy and we have DNS plus service, so SRV etc. available

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    PBX skilled programmers Đã kết thúc left

    Please apply, we need someone who is expert in PBX, we need to resolves our issue regarding this. We will send the credentials to the right candidates. If you cannot work today, do not apply! Thank you

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    ...in Germany. We have multiple Solutions and now are expanding to telecomunications security. We need developers with experience in VoiceXML and JS, as well as knowledge in Asterisk. We are building an IVR system with dynamic and automated functionalities and offer Jobs with contractual regulations for at least 3 1/2 years, after hiring! We pay on a

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    Top Programmer for Source Code Đã kết thúc left

    To find and change things in source code of big PBX w. WebRTC, with little or no comments. Fix problems in software made by different people and add comments to source code. All documents to explain functions of system, are available. You must be able to work with Teamviewer.

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    Set up office VPN Đã kết thúc left

    I need to install my draytek vpn router so that I can access my office phone from my mobile and desktop devices when I am outside the office. I have a yeastar s20 IP PBX.

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    Asterisk integrate with Hubspot Đã kết thúc left

    hi duke, i need a team to integrate Asterisk call center with Hubspot to have some feature bellow: - Sale can use Call feature on Hubspot via Asterisk. - Can use feature Click to Call. Please show me the solutions (brief, draft with your paper) to continue!

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    ...dtmf or voice dial commands based on what comes in the sip header. the sip app will register to a sip sever who will send it numbers to be dialed in the header When the sip server sends a call, the sip app will issue dial command via the headphone speaker. (to phone 2) if it hears a acknowledgement that the dial was received, wait if no acknowledgement

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    ...in Germany. We have multiple Solutions and now are expanding to telecomunications security. We need developers with experience in VoiceXML and JS, as well as knowledge in Asterisk. We are building an IVR system with dynamic and automated functionalities and offer Jobs with contractual regulations for at least 3 1/2 years, after hiring! We pay on a

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    ...straight forward. Just a PBX with 4 extensions, and a trunk. All works well. This is shown in the diagram as 'Before'. One SIP phone however is very special. It doesn't have a transfer button. So no call transfer can be initiated on the phone. It should be possible however to initiate a transfer via DTMF tones. The current PBX (main PBX) doesn't support...

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