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    18,524 android sip client asterisk pbx công việc được tìm thấy, giá USD
    Project for Vu L. Đã kết thúc left

    Hi vule83, mình đang có dự án Odoo CRM tích hợp tổng đài voIP sử dụng Asterisk 11. Bạn đã từng làm dự án nào tương tự chưa? Nội dung cụ thể thì cho mình số dt của bạn để trao đổi cho nhanh nhé

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    Chào các bạn, hiện tại tôi đang rất muốn kết nối vtiger opensource 7 với tổng đài assterisk, nhưng tôi không biết làm việc này, rất mong các bạn hỗ trợ tôi kết nối giữa 2 ứng dụng này, tôi sẽ trả phí kết nối cho các bạn. Trân trọng cảm ơn!

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    Project for nttranbao Đã kết thúc left

    Hi nttranbao, I'm a newbee asterisk. Now, my project are doing function monitor video call with asterisk. Do you do that function? If you can, you can send email for me ( [đăng nhập để xem URL]@[đăng nhập để xem URL]). Thank you :) Hy vọng anh biết tiếng Việt. vì tiếng anh của em không tốt lắm. Em mới nghiên cứu asterisk và hiện tại dự á...

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    Mình cần hỗ trợ config Elastix, inbound, outbound và sip trunk

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    change logo elastix PBX Đã kết thúc left

    Tôi cần thay đổi logo tổng đài elastix I need Change logo elastix PBX

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    Install and configure FreePBX 14 on Digital ocean and setup, needs to support commercial modules...

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    Gsm voip gatway 6 ngày left
    ĐÃ XÁC THỰC

    I need an Android app. I would like it designed And Build Android application make mobile phone act as GSM to voip gateway To connect with Asterisk PBX We needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. It aim to make p...

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    Our project consists of two parts: 1) Select the correct customer inside POS automatically whenever the ...Odoo POS, and the system then knows that this delivery man has delivered the scanned orders. We will provide an Odoo v9.0 server where we will also have installed a Freepbx asterisk server along with the OCA connector-telephony plugins installed.

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    need part time permanent remote freelancer 6 ngày left
    ĐÃ XÁC THỰC

    Hi. we...one on one audio, video sessions just like skype. this app has: audio, video, screen sharing, conferencing , ios native and android native apps. it is built on webRTC, Node.js , angular.js , asterisk, xcode, callkit, android studio. we need someone really expert in above skills. and you should have some apps ready similar to above skills.

    $25 - $50 / hr
    Nồi Bật Đã niêm phong
    $25 - $50 / hr
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    Need a PBX system made 5 ngày left
    ĐÃ XÁC THỰC

    I have a company with multiple phones and would like to keep better track of conversations. I need a simple system that when called redirects to my company numbers. I also need it to record all conversations also.

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    Country Sales Manager UK 5 ngày left
    ĐÃ XÁC THỰC

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

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    Sales Representatives for SaaS in the UK 5 ngày left
    ĐÃ XÁC THỰC

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

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    Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDr...integrated into pipedrive are below. [đăng nhập để xem URL] Freepbx is an ASTERISK based phone system. Need to know how long this will take. Must have prior experience with Asterisk based phone systems.

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    asterisk job 5 ngày left
    ĐÃ XÁC THỰC

    asterisk person to work on an existing product to fix and update the product knowledge needed: PHP Asterisk database

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    LinPhone Development Project 5 ngày left
    ĐÃ XÁC THỰC

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    Low bandwidth voip solution by asterisk. 4 ngày left
    ĐÃ XÁC THỰC

    well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should

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    I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Địa phương
    $9 / hr Giá đặt trung bình
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    I need confirmation free PBX raspberry + 3g usb modem k3520 .

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    ...destination and duration of these expenses (local / national / international) The detail for these reports is found in the CDR's (call data records) obtained from the customers PBX. We have a self-developed, web-based cloud platform (the back-end is on a Dot net platform,. The front-end is based on HTML, Java Script and CSS) that is providing this basic

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    Freepbx and Asterisk settings 4 ngày left
    ĐÃ XÁC THỰC

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de

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    Only for gurus familiar with Bitrix who can setp Bitrix application to work with our PBX Need to use Telephony REST api

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    Personalisation LinPhone Security & Design 3 ngày left
    ĐÃ XÁC THỰC

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    a2billing set up 3 ngày left

    a2billing setup, calling card, PBX, Auto dialing, SMS

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    ... We can discuss any details over chat. Below a summary.. I have a IP PBX with 2 sets of 10 numbers. Set nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer

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    ...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....

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    ...phones system and microtik. i had another IT Specialist setup a mikrotik router in each one of my locations. i have 2 locations. They are connected through sstp vpn. i have a 3cx pbx in location 1. location 1 has 6 phones and location 2 has 6 phones. Network access to both locations from one another are working perfectly fine. i am having trouble with phones

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    CRM (php) Integration with 3CX 2 ngày left
    ĐÃ XÁC THỰC

    Hi, I have Php house made CRM, want to integrated 3CX PBX VoIP with CRM, Adding extra field in the customer profile (email) and generate report. More information are available in the attachment.

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

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    Customization of LinPhone SIP / VoIP Client 2 ngày left
    ĐÃ XÁC THỰC

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    10 lượt đặt giá
    VOIP yate call pass 2 ngày left
    ĐÃ XÁC THỰC

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    app development 2 ngày left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    integrate bitrix with asterisk system -

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    ...modify those. Below are more detailed descriptions of the api/ and frontend/ directories. Locations that you will be mainly focussing on as you develop are marked with an asterisk (*). Backend API (api/) api ├── Dockerfile # Description of Docker image ├── [đăng nhập để xem URL] # NPM dependencies ├── spec # The backend tests

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    Wordpress Plugin for Phone Call 1 ngày left
    ĐÃ XÁC THỰC

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

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    Actionscript 3.0 SIP 1 ngày left
    ĐÃ XÁC THỰC

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    PHP SIP client 1 ngày left
    ĐÃ XÁC THỰC

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project 1 ngày left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    FreePBX support 1 ngày left

    ...FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from a paid PBX (switchvox) and move to a

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    asterisk expert needed 1 ngày left
    ĐÃ XÁC THỰC

    Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

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    Create SNOM Phone Dial Plan 1 ngày left
    ĐÃ XÁC THỰC

    I am looking for a ...10 digit local number we want 9XXXXXXXXXX sent to the PBX as it will strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    SIP To SIP register over Masquerade VPN 11 giờ left
    ĐÃ XÁC THỰC

    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [đăng nhập để xem URL] or yate or besip [đăng nhập để xem URL]

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    add sip trunk to elastix Sắp kết thúc left

    add sip trunk to elastix i have sip trunk from STC

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    We have a company based in kuwait and we want to set a " Autodial" management system for our sales team of 5 pe...customers which is assigned by the system to each extensions call will be thrown to them as soon as each people keep there telephone down. We have following things 1.) GRANDSTREAM PBX with telephone 2.) Servers if required. 3.) VPN set up

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    configuracion de internos y de central asterisk

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    Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem

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