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    10,422 softclient sip tugasan ditemui, harga dalam USD

    I'm looking for experienced web developer with WebRTC experience building dialing systems from scratch. We have a cluster of multi-tenant PBXs (VitalPBX) the new dialer must be integrated with. We expect this dialer to allow agents to place/receive calls, hold calls, transfer calls and any other normal call centre operation features to be implemented. We are open for the technologies you would like to use, e.g. it can easily be React.js with JSSIP library, or Vue with Sipml5 etc. It's would be also nice to stay in touch with developer for further development and some support in case we need something to know or to get be able to consultate with developer per hour/project basis. Minimum experience required: 2+ years web dev expertise with solid Javascript knowledge, WebRTC an...

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    Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards

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    Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards

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    Buenas tardes, en estos momentos tenemos un Webphone desarrollo con la tecnologia SIPML5. El webphone funciona de manera correcta y esta desarrollado en javascript con .net. El inconveniente se da que a veces se suspende la sesion del explorador y necesitamos evitar eso. Entiendo que necesitamos un experto en javascript que nos ayude a ver este punto y que haya desarrollado un webphone antes.

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    We are changing from one SIP provider to another one, and need configuration update. All the configuration is in place, and just need to change to the new one.

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    Segera
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    5 bida

    need only software for Cisco Autonomous AP IOS Software for two access points (Aironet 1600 model AIR-CAP1602I-E-K9 and Aironet 1700 model AIR-CAP1702I-E-K9) and SIP software for telephones cp-7900 , cp-7821 cp- 8841

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    Trophy icon Logo Design 1 hari left

    I am looking for a logo. It will be for a grocery store and marketplace. Name is "Let's Get FRESH" Grocer & Marketplace. The grocery store in locating in the Pickering Passage and Spencer Lake area. This is Washington State so Mountains, trees, ocean, shellfish fish etc. Font should convey be organic, natural, fresh, community, etc. Colors Sip of orange, bamboo beach, etc.

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    1169 penyertaan

    We are working with a telecommunication vendor that exposes a phone number and does SIP forwarding into our system. We are using Asterisk today and seek to migrate from traditional IVR to connecting to our dialogue engine, exposed behind a websocket server. We are not bound to Asterisk if there are better components for receiving SIP calls and converting to websocket. Please reach out if you think you can help.

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    6 bida

    Right now we have 1 cloud FreePBX server (Ubuntu) at internet, there is Strongswan IPsec tunnel to the SIP provider (connected) but now we need set SIP trunk just over this IPsec, anything else at public IP has to be working...

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    ... 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 versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/ calls. Functional flow 1) Calls originating will send to whatsapp gateway 2) whatsapp gateway co...

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    Vici Dialer Tamat left

    1. Installation & configuration Vicidial 2. Predictive/Manual dial option press 1 campaign 3. Able to make 300 concurrent calls at once configuration & Firewall for ssh Inbound - Call menu - avatar DID groups user creation phone creation Music on hold only with avatar and press 1 then with firewall configration phone and user creation SIP Trunk and carrier configurations

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    Configuration of DVG600s to send calls, and configure cloudfonica pbx to send calls to the gateway the gateway will send the call to PSTN. only 1 channel FXO. Please consider this requeriments: Cloudfonica PBX should send SIP Invite packets to Dlink Gateway. If FXO lines form PSTN are not working, calls can not go through.

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    I am trying to add Twilio SIP to my Snom 870 Phone. Can you help?

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    Trophy icon Design Cake Box with Style 9 jam left

    Hi, I would like to sell this famous apple pie called "Apple Strudel" in a unique and eye-catching box. The recipe of it has some main ingredients as Apple, Pine-nut, Raisin, Cinnamon, Honey and sip of rum and sugar powder on top. Would be cool to see on the box: - the logo of my shop in two color - a famous mountain: 3 cime di Lavaredo, that remind where i'm located so probably tourist like to get the cake box as holiday gift. Show me your stile :) in Italian the cake is called "Strudel di Mele" Box size: Length 38 cm Height 6 cm Depth 10 cm

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    18 penyertaan

    i need to configure D-Link DVG-6004S fxo to be able to dial out calls received from sip through pstn lines

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    Need an EXPERT to setup and configure Sangoma FreePBX server, users, extensions, VM, etc. Connect to SIP trunk. Test, train, and maintain. Less than 20 users initial roll out.

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    25 bida
    Twilio setup Tamat left

    Need to connect Twilio SIP Trunk to Field Edge

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    20 bida

    I just started new business. I need experienced VOIP expert only accept he/she who have rich experience in freepbx and asterisk and sip field. best regards

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    21 bida

    Advertisement Plot- A guy comes from the office and seems to be tired and stressed. Later he goes out for a walk and finds some people enjoying cole over chit-chat. The guy goes to a vending machine and put a coin in the machine and get his can of coke. The moment he had his first sip. The guy goes into his imagination and sees people dancing in joy. Also, people who are going home from work are dancing. People spread over the street and enjoy every moment. The guy who drank coca-cola sees people inviting him to join. Being shy he says no. Still, people are encouraging him to join but he feels shy. Then in the atmosphere of such happiness, the guy hears the sound of a horn. And he comes out of his fiction world and find himself in the middle of street. And he gets off th...

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    customizer firmware Hikvision .Dav add additional new features, experienced professionals only. work on SIP call protocol, enable auto answer in door station customizer firmware Hikvision .Dav agregar funciones nuevas adicionales, solo profesionales con experiencia. trabajar sobre protocolo de llamadas SIP, habilitar auto respuesta en estacion de puerta

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    I want a SIP trunk to connect to softphone and be able to change my phone number/caller ID to whatever I want whenever I want with ease, and call from whatever phone number I want to display on other side.

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    3 bida

    We are building a simple business communication app for entrepreneurs. It is something similar to Google Voice. We have built most of the features except certain aspects of the call features itself. Who we are looking for: 1. You have a proper understanding of voice communications (SIP call flows, WebRTC, SIP over WebSockets, and so on) 2. You have more than 6 years mobile app development experience. 3. You have more than 2 years experience implementing WebRTC with Flutter. 4. You are comfortable with the GetX package for routing and state management. 5. You are able to write platform-specific code (Kotlin and Swift) if necessary. 6. You have experience implementing Callkeep (CallKit and Android Connection Service) 7. You speak English fluently. 8. You are availabl...

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    I have the latest ASTERISK and FREEPBX set up on a hosted Linux machine. I am able to receive calls via the three sip trunk providers, but I am unable to make outbound calls on these sip lines. I need someone to simply diagnose the settings on the FreePBX interface, and adjust them so that they can dial out. For this, you will be provided a login to the FreePbx Admin interface, where all settings are available. You MUST have previous ASTERISK and FREEBPX experience at a system administration level.

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    Build Website Tamat left

    Website Requirement Objective: ❖ Responsive and SEO friendly website on React js and Node js Including Code). ❖ Milestone/SIP Calculator and Planner. ❖ Rotating homepage banner. ❖ Social logins through Gmail ,Facebook and LinkedIn. ❖ Easy to manage services offers, Predictions fromBackend/admin. ❖ Online Chat. ❖ Customer /Partner sign up and login control panel. ❖ Email to friend/social sharing of services. ❖ Add / update/editPredictions. ❖ User friendly Navigation ❖ Payment gateway (TBD). ❖ Newsletter signup ❖ Coupons and and Integration of Referral system points ❖ Upload documents feature ❖ Market,Finance and Currency news section(Help of API). Stakeholders: ❖ Admin ❖ User/Investors ❖ Partners/Employee Website Module and Behaviour: ❖ Dashboard. ❖ Invest Plan Management (Need help ...

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    Hi, I have a Ruby (3.1.1) project that needs to send calculated UDP payloads via SIP (RFC 3261), an example payload being: Sending via Ruby is not essential, I can call command line clients (Centos 7 preferred) I will provide access to a server, I need a developer to create a script that can send the above to a host, and advise me how I can put in my own payload body. I hope this is enough information, any questions please let me know. Thanks

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    Hi , We have AtlasIED PA1001G atlas sound amplifier works fine and connected with telephone system analogue and all speakers works fine , now we have added VoIP phones that needs to add algo device to connect to it the algo device is configured with sip and ready . Final step need to connect algo to atlas sound amplifier ? Anyone can do it ? Please advice

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    FIX Linphone SIP Audio Issues after upgrading from 4.4 to 5.0 (Cordova/Ionic) Device issue is Lenovo M10 (some models do/dont work) THe far end can hear audio but the Speaker of the device blanks out audio. it did work on 4.4 then another Model device Didnt work on 4.4 so it might be a config adjustment in our linphone setup

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    5 bida
    Sip scanner Tamat left

    I need to make a sip scanner in cc++

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    Trophy icon revamp/modernize my logo Tamat left

    looking to revamp or modernize my logo. The font is boring but I like the house with the girl in the middle concept. I do custom wooden decor, paint and sip nights for groups and custom t-shirts and mugs.

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    Dijamin
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    48 penyertaan

    PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak uydurabi...

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    Hi I am looking for a developer with extensive experience in Asterisk to build a call center based solution over SIP. Only experienced developers who have worked on this earlier should bid. If you have already developed complete solution for it, we might also consider it.

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    Need softphone application for desktop (all Windows Operating systems). The softphone will work on LAN, there is no internet connection or ISP. Feature of application only Voice call including Call holding, conference call, call Forward. Some examples are 3CX, Micro SIP, PC- telephone and Magnet Talk Softphone.

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    5 bida

    Dear Candidates, We are a telephony application solution provider looking for developers who are proficient in Linphone SDK to develop mobile applications for IOS and Andriod. As a cloud telephone service provider, this SIP mobile application shall register to our SIP server using SIP and TLS communication. This is SIP mobile application to be precise, it is designed to support basic phone calls such as incoming and outgoing calls. The candidate must consider that this application is to build on the SDK and it has to be configured particularly push notifications, VoIP notifications with the call kit. Bear in mind, that we have limited time and we must make sure that the application performs well as a basic telephony system such as incoming calls management...

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    Looking for help in FreePBX setup, f...attached file () A. FreePBX is installed on Raspberry computer, B. Raspberry has input 3 lines, and 1 output 2x VOIP Landline 1x SIM Card Number (Usb Dongle) and 1x ZOPIER sip C. Always on connection FREEPBX automaticly pickups call and start IVR menu Select 1, Select 2 (welcome msg and select menu) D. When selecting 1 or 2 - all calls are forwarded to ---> SIP on ZOIPER Necessary: -Working Hours (after 4 p.m. there is another greeting) -Music on Hold (after selecting from menu 1 or 2 you will hear MOH until pickup call by SIP ZOIPER) -Fail2Ban -Backup on 2nd card Info: USB Dongel works without problems Raspberry works without any problems Card #1 is 64 GB Card #2 is 16 GB I...

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    Hi Felipe M., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    Need peer to peer VoIP application for desktop (all Windows Operating systems). Feature of application only Voice call including Call holding, conference call, call Forward. Some examples are 3CX, Micro SIP, Zoiper and IP telephony.

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    5 bida

    VoIP/Sip client Android APP first of all the app must be fully encrypted. this app must me done with the SIP protocol standards. Basic Features: Incoming Calls Outgoing Calls Mute Calls Hold Calls Transfer Calls Integrated Call History Contac List change CallerID before calling or fixed optional: Send random caller ID before call. Advanced Features: voice/speech changer integrated to the calling app Multiple SIP accounts setup SIP Account information hidden after initial setup DTMF (telephone tone) support using SIP INFO or RFC 4733 Provisioning via QR Code SIP TLS transportation SIP UDP transportation SIP TCP transportation Audio codecs: G711, G729, GSM, G722

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    18 bida

    This project requires a technician that is very familiar with a Mitel Office 250 system and has experience installing wireless SIP phones on a commercial WiFi network. If you are not familiar with the Mitel systems, please DO NOT BID. We purchased a brand new Mitel Office 250 system and already had previously attained 2x Mitel 5624 v2 phones. Mitel has confirmed that everything was compatible and we also purchased the required licenses. We require you to get the two 5624 phones working on our system. I am a system Administrator so I am not having luck making them work. Everything else is currently fully functional. Here is the current status: Phones were proven to work fine when installed on a “Simple network with no VLANs” but I cannot get them to work on our curren...

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    Necesito conectar una pequeña planta IP con Gateway GOIP GSM 8 Canales, para poder hacer y recibir llamadas desde las extensiones SIP a través del Gateway GSM

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    12 bida

    Hello I am looking for someone who is able to buid video door phone with raspbery pi using sip or webRTC if interested please conet me thanks

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    english >--------------------------------------------------------------------------------------------------------------- -- SIP client as APP for Nextcloud with JSSIP library. The following functions should be supported: To dial a keypad 0-9 *,# Answer and hang up with DTMF transmission. 1. register extension 2. make call 3. accept call german >--------------------------------------------------------------------------------------------------------------- SIP Client als APP für Nextcloud mit JSSIP library Folgende Funktionen sollen unterstützt werden: Zum Wählen ein Tastenfeld 0-9 *,# Annehmen und Auflegen mit DTMF Übertragung 1. Nebenstelle registrieren 2. Anruf ausführen 3. Anruf annehmen

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    Setup Kamailio Tamat left

    We need help installing a Kamailio SIP redirect server. See specification in PDF Thanks /Dan Erik

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    Build WEBRTC SDK for developers to Connect + Documentation need to build a WEBRTC SDK (gateway to ) but not entirely sure how to build a SDK so users can use our SDK / Examples to code into their app shell. android/iphone/electron/browser. etc etc

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    Want to you assist me to configure the Desktop IP phone for SIP to be registered

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    Hi, Thanks for reading. We have a project that has a number of elements which we are hosting on a web server. The elements include: 1. Asterisk ip PBX 2. Kamailio SIP Server 3. A database for the above 4. An MQTT Broker 5. A web server to support all the services. The system is an emergency call system fort he elderly and we need to be sure we have resilience We need a person to review from the outside and make recommendations on how we make the system scalable and robust. You will need knowledge in: - Asterisk - kamilio - PHP and web servers - MQTT Brokers This is a short project but we see further long term work in helping us as we add more features. Thanks for reading

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    Profissional devera efetuar a configuração da máquina virtual 3CX na Amazon AWS adicionando ramais a partir de uma base de usuarios Active Directory (AD), adicionar conectividade ao Microsoft Teams Voice e Conectar um Tronco SIP para chamadas publicas de entrada e saida.

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    Install kamillio on cloud VPS to act as proxy sip server To serve few system already installed such as issabel, grandstream, yeastar So they can operate remote extensions without the need of natting or port forward on the router of the IP system

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    Hello I am seeking for someone who build video door phone with raspbery pi using sip or webRTC if interested please conet me thanks

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    1 bida

    Hello I am seeking for someone who build video door phone with raspbery pi using sip or webRTC if interested please conet me thanks

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    1 bida

    We are seeking firmware development for the nrf9160 SIP. See own document for brief intro

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