Wengo sipcông việc

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

...project is to create a SIP to WhatsApp gateway that will allow voice calls to be passed from SIP to WhatsApp, ensuring that the call reaches its intended recipient. The development platform or operating system is not a concern, as the project can be completed using either the Linux/Windows WhatsApp executables or the Android/Windows Phone mobile versions of the application, regardless of the version number. The gateway must return appropriate call error codes to the SIP backend, such as CALL SUCCESS, BUSY, UNAVAILABLE, etc. To determine the success of the project, we will choose the solution that consistently triggers successful SIP/WhatsApp calls. Step-by-Step Process: Incoming calls will be directed to the WhatsApp gateway. The gateway will then convert...

I need someone to build hardware prototype GPS device using module NRF9160 SIP (Nordic Semiconductor). It needs to send data to server using internet from SIM card, also battery level and accelometer data from sensor. For more information please contact me.

Setup docker/ kubernetes based call center / contact center Inbound and outbound calling Experience in setting to similar contact/call centers, using WebRTC Experience of administering SIP / VOIP

Hi I need a sip trunk for calls acd USA. I have almost no asr - but all the acd is legal and approved by the customers. If you are dealing with the subject and know of such a supplier or are one yourself, contact me. If you have no experience in this matter then please do not contact me

Hi I need a sip trunk for calls acd USA. I have almost no asr - but all the acd is legal and approved by the customers. If you are dealing with the subject and know of such a supplier or are one yourself, contact me. If you have no experience in this matter then please do not contact me

My interest is to configure voipbuster by sip trunk in issabel, in such a way that I can send the callerid from issabel

Setup Vicidial Full configure with Telnyx or Pilvo VICIDIAL full setup with SIP trunk - we provide server.

I need a solution to resell voip. my provider gives me sip trunk then I want to resell ports to my customers that uses asterisk PBXs

Preciso de um desenvolvedor REACT/ e NODE.JS com bastante habilidade de agilidade. Além disto, preciso de um profissional que já tenha construído sistemas com protocolo SIP. ATENÇÃO: APENAS DESENVOLVEDORES QUE FALEM LINGUA PORTUGUESA.

I have backend and Janus Webrtc gateway with SIP plugin. On HTTP request to backend API it handles Janus commands and returns session_id, handle_id and token. I need html and javascript to continue from where to initiate new call via SIP plugin or answer incoming call (Janus running in HTTP mode). I will pay 50 USD.

We are looking for someone who can install on our server and make it have astersk setup and working. Just need audio call option

I am in need of a grant to assist in the growth of my business. I am a small business who started in 2019. We offer paint and sip, crafts and other types of art. We have classes in our main location and travel to resorts, wineries, VRBO and other places. We have parties for bachelorettes, birthday’s, and any type of event. We need additional funds to expand the business in additional areas and locations, hire more instructors and add to the opportunities I have written a few grants but have been unable to secure one. I have my business plan and information ready, I just need assistance on putting that into a grant proposal.

Project : Dongle Panel Function : web-base hosted VoIP to GSM gateway is accessible from anywhere on any mobile device, it helps to terminate/originate SIP calls traffic using 3G Modems/Dongles and an active USB hub connected with your Linux PC. With the features of exquisite enclosure and high performance, specially designed to maximize cost . Dongle : meaning Huawei usb modem with sim card attached to Freepbx or issable End-User • Same Panel (As Attached Pictures) • Add Call Feature via panel • For all functions I supposed to choose which dongle i need to use . Admin-User • Control All users (Add,delete,rename,change password,list of all users) • Payment method : PayPal & Stripe (2 options will be monthly

potřebuji do svých stránek udělat sip klienta. Používám laravel, php a javascript. Takže nejlépe v JS. Potřebuji to promtně turbo rychle

...and to preserve the heritage that has been passed down for generations. At Qahwa, we believe in quality over quantity. We source only the finest Yemen Mocca coffee beans and roast them to perfection, capturing the essence of Yemen in every cup. Whether enjoyed in our coffee shops or at home, we strive to provide an exceptional coffee experience that honors the rich tradition of Yemen. With every sip of Qahwa coffee, we hope to transport you to the mountains of Yemen, where the aroma of coffee fills the air and the beauty of its heritage inspires us all. Join us on our journey to preserve and share the legacy of Yemeni coffee, one cup at a time...

Diseño y desarrollo de una aplicación Aplicación móvil OTT Windows – Linux – Android (todas la posibles versiones) -IOS, iphone, - IPN(las diferentes versiones) con las siguientes funcionalidades: - Cliente SIP para VoIP, con la opción de elegir entre servidores SIP - Cliente de chat (REST API), el chat debe ser capaz de enviar y recibir Audios, texto, emojis, imágenes, vídeos, ubicación y archivos - Soporte de videoconferencia - Notificaciones PUSH - Soporte de cliente VPN

Looking for a sip softphone for windows desktop with g729 codec ,call transfer and conference option. Should work behind nat. Stun and rtp ports should be auto filled and static. We will build with basic options now and will upgrade more , so bid with your best budget.

We have an extension which is registered to the asterisk server.. but when making a call the Asterisk returns 401. The task is to find out: 1: why this is happening 2: what we need to do to fix it 3: Make test call to prove fix We will only pay for a fix..

I need to Configure SIP trunk between the billing platform MagnusBilling and the MultiTenant PBX. I need to set it up so that all tenant PBX's are able to call out from the PBX and get billing individually by the MagnusBilling platform.

Here are goals for the modification, 1. Modification for both MacOS and Windows 11 platform. 2. Provide installation packages for both MacOS (.pkg with cert) and Windows 11 (x86 - msi/exe). 3. Rebrand Blink to our product brand (2 brands). 4. Modify login to use Web Services, this web services will be provided by us. a. Web services will providing SIP credential and phonebook. b. Web services will be using encrypted messages. 5. Removing unnecessary GUI and functions. 6. Synchronize contact/phonebook with cloud Web Services. 7. Change the color scheme of the app according to our specification. 8. You will need to handover the codes. 9. You need to prove the codes are working and can be compiled using virtual machines setup by us. 10. We will provide the MacOS and Windows en...

...business logic Our VoIP Arch -> Connected to multiple sip endpoints for each users on their account. I need some one who can help me setup an asterisk on linux machine probably in some better datacenter such as aws or azure that could run behind an proxy public ip for client server connection. And than use php agi to develop an sdk that could send and receive rest api and xml commands for controlling ivr dialplans for different users that would be connected in our subscription based billing system to run use the api to connect different sip channels they add in their account and use their app and websites to connect and control the ivr diaplan in their ivr. We also need to make sure that users on our subscription are able to add sip channels they own and ...

Need create notification for ippbx support ios and android (React Native) want to make notification to support app wake When the screen is turned off or the screen is locked when someone calls through the sip application, both ios and android

Need some assistance to getting our app to work Incoming and Outgoing calls. demo like ctxSip

* Modern Looking Softphone App. Flutter language to be cross platform, SIP coding, should include: - User Registration form; - Keypad to enter phone number; - Access to phonebook; - Separate page to send and receive SMS/MMS; - Show Balance; - Recharge Balance; - Group call; - Low Battery consumption for background activity of app; - Add users through phone number; - Indicator if other user is online/offline; - Ringtone selection; - Voicemail; - Call waiting; - Users can add multiple phone numbers after purchase; *Multi-Page App includes the following pages: - Softphone (keypad page); - Users page (where users can see other online/offline added users/add users, chat through SMS/MMS, or create groups to make group calls); - Wallet page (shows balance, users...

require sip trunk call to china with randomly display CLI

Create a small demo project in Delphi 10.2 Tokyo or higher that uses the BareSIP libraries () to create a basic SIP client that registers a SIP extension to a SIP server (FreePBX or 3CX for demo purposes), makes a call to an extension, sends a WAV file as G.711u audio over the call, and hangs up the call.

Sip panel engineer, drafting. Need stamped drawings. So the plans can be submited to manufacture and city zoning committee Structurally insulated panel design Structal insulated panel engineer or architect. Familiar with SIP panels.

Deploy a cloud voip solution in aws configure application for multi Tennant pbx deploy sbc and configure and test

The goal of this project is to create a step-by-step guide for configuring Kamailio (a free and open-source SIP server) to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown as it will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Kamailio or Martini Security's offerings. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Kamailio with STIR/SHAKEN. Martini Security offers a certificate enrollment

...using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN (). Martini Security offers a certificate enrollment client called Olive () which can be used if FusionPBX/FreeSwitch has working STIR/SHAKEN support built-in. If it does not have native STIR/SHAKEN support, the Martini Security STIR/SHAKEN signing and verification server called Vermouth () can

...will provide a .pcap file captured with tcpdump. The file captures VOIP calls made on our network. You will write a command-line utility program that reads the pcap file and outputs the RDP audio streams into .au files. The audio is in 8000hz PCU format generated by SIP telephone calls. For each SIP telephone call you will provide THREE audio files. One file will contain the RDP audio for each separate side of the conversation. The third audio file will contain BOTH sides of the conversation. In addition, for each SIP conversation, you will provide a data file containing JSON text, showing the TIME and DATE of the call for the PACIFIC timezone; also TO and FROM fields showing the names of the caller and person called, if these are available in the ...

Early-stage, Midwest, startup gearing up to produce SIP wall, floor, and roof components.

Installation eines Jitsi Meet Servers auf bereitgestelltem Ubuntu Server. Zugangsdaten stehen bereit Grundkonfiguration Jitsi für Browser und App-Nutzung Nach Rücksprache: - erstellen eines SIP Accounts - Setup FreePBX mit IVR Einwahl

Telecom billing and routing software ASTPP installation and setup for community and enterprise license - Full setup with sample customer. -Gateway integration -SIP proxy. -Reseller account. -Payment gateway etc.

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

I need a cloud FreePBX system to manage my sip numbers. In the meantime, I need to use database to store all of the CDR (including incoming and outcoming, etc.). Moreover, the instant incoming calls need to show to a webpage on my front-end app, and the response from the app (such as answer, hang up, hold on) will send back to the Asterisk.

Buscamos un perfil que cree contenido referente al contact center, debe tener conocimiento de dicho sector, customer experience, employee experience, software de contact center, trunk sip, centralitas, departamentos de contact center, etc. Que sea de idioma español nativo. Conocimientos del contact center. Queremos que este con nosotros mensualmente, hay mucho que redactar al mes, artículos, cursos, grabar podcast, pdf de contenido para su descarga. Tenemos 7 lineas de negocios y cada una de ellas es de una rama de contact center.

...include integrations with external APIs and the ability to create and sell products through a WordPress module. In addition, the software should include stand-alone modules for live chat and SIP communication, a learning management system, and appointments. There are several additional features and functionalities that are outlined in the attached document. It is important that developers carefully read and understand the entire document before making a bid on this project. The document includes details on the Master SaaS, Invoicing, Safe Mail, Modules, Stand-Alone Modules, Templates, Accounting, Chat and SIP Client, and Onboarding Wizard, as well as using external APIs and storage. Developers should have experience in building software solutions with similar features an...

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

I need someone to setup a simple javascript control of SIP (phone calls) using I have a SIP trunk with Telenyx.

i want a website for investors it should contain sip calculator, online investment and some thoughts

sip softphone development with source code and CRM integration

- Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP An standard SIP client running on a separate android phone will have to connect to the SIP server that was developed on the other android phone via wifi and an incoming and outgoing call would have to be demonstrated. Note: please read first & check android condition, if need root version use i thinking it. First you show me demo then add money freelancer website.

Hello, I would like you Caller Id asterisk sip that can be used anywhere in the world. Thank you

I use 3CX startup. I want to add a sip trunk for calls in Belgium. I have problems with provisioning my Yealink W76P (W70B & W56H) and use it as a router phone.

We need to finalize the goautodial v4 configuration on a cloud server: 1- Finalize the configuration of the sip trunk 2-Deploy a campaign (test) 3-Configure incoming calls