Softclient sip työt

I have a Node JS SIP module that I have to work as regular java. I need an EXPERT in Javascript to work on it with me. By EXPERT i don't mean someone that knows basic JS. I mean an expert

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...

We are looking for feeepbx with Gsm gateway setup for incoming and outgoing calls you should have experience on FreePBX and make FreePBX secure . And we want to use WireGuard on same server to connect out sip extensions and gateway as trunk

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

...completion the coding. Looking for a Senior Java Developer to finish coding XanticTek TASS application. Must have skills: • More than 5 years of professional Java 8 (or higher) development experience • Understanding of maven multi-level projects • Knowledge of Java Swing (for the UI) • Writing SQL queries for Postgres DB • Good level of Linux knows how Nice to have: • Deeper knowledge of VoIP and SIP protocols If you don't meet the “Must have skills” and you don't have the curiosity to learn “Nice to have” there is not making any sense to open discussion with us. Category: Software Domain: VoIP (voice over IP) Function: AntiSpam System for VoIP & SMS traffic Documentation: all Features are extremely well pr...

We have a problem with a Cisco ATA-192 that will not register with our Asterisk server. With tcpdump we can see REGISTER attempts from the ATA and a 401 response being sent back, but nothing further. We suspect it may be a NAT issue. The remote end is behind a broadband router and is NAT'd. We need a network expert that also understands Asterisk and SIP. Also, ideally how to configure a Cisco ATA-192.

Hello I am looking for designer to perform a SIP design home about 1500 sqft. on a down slop hill of 40%. 24x56 ft. athe picture will give the idea.

I would like to develop a SIP/VOIP application for mobile and desktop which will be used in fusion pbx

I need to monitor my siip server. The correct freelancer will suggest which app/service is best to use for this.

...to access web interface - Billing System for subscriptions that integrates with our credit card processor, Stripe - Reseller Support - Auto Dialer for Client Call Centers, with admin interface and exportable logs, easy import for dial lists. - Automatic Failover and Load Balancing for Redundancy across multiple physical servers / datacenters Current Features we do have: - SIP/PJSIP - Find-Me/Follow-Me - Voicemail to Email - Call Waiting - Call Whisper - Do not Distrub - Extension to Extension Dialing - Call Forward - Call Parking - Call Transfer - Caller ID Screening - Successive or Simultaneous Ringing - Three-Way Calling - Voicemail - Custom Hold Music - Dial by Name Directories - Multi-Company Profiles for ...

Dear everyone I have one Call center platform. It worked well some days ago. But suddenly, freepbx is down. So i restored previous backup file. after restored, my call center platform worked again. But some modules doesnt work correctly. I tried to contact previous developer to solve this problem. But due to sick, he can not work right now. so I need someone to so...But suddenly, freepbx is down. So i restored previous backup file. after restored, my call center platform worked again. But some modules doesnt work correctly. I tried to contact previous developer to solve this problem. But due to sick, he can not work right now. so I need someone to solve these problems. the call center built with laravel. someone should have rich rich experience in laravel, sip, freepbx, webrtc. Bes...

A copy of the poster is attached, 5 poster a week $20 per poster. Please i am not asking for too much stories. This are all the poster i need. Exotic Painting Paint & sip Booking; minimum 10. group 10% voucher food promotional ladies night @hollywood entertainment. Lunch special African food birthday parties. Book ur event. Veunue Hire Spicy Saturday event. Every Saturday Every Friday Karaoke like us IG : FB website. @myplacebarperth

we have FreePBX server and we need to configure the IVR dial Plan with setup SIP trunk inbound and out bound

Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.

To invite everyone around Little Alden and Spring Lake Sunday December 11th, between 1pm and 4 pm At Cindy and Rod's, 3296 N Little Alden Lake Road, 218-391-5815 We'll have Fires going...we'll have Hot Chocolate to sip on and all the ingredients for you to make Smores ! Bring any Beverages that you'd like !! Sit around the Fires, get Cozy in the Bunk House or Relax in the Cabin Wear Some that is all Christmas...A Hat, a Sweater, some Socks...We'll try to get some good Pics to put together in a collage !! We hope to see you Sunday ! We just need a basic flyer to distribute around the lakes and to email. Use some Christmas colors...make it fun... I am looking for an inexpensive flyer that is professionally done..." not in my handwritng " Than...

...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 matter 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, that triggers successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to the WhatsApp gateway 2) WhatsApp gatew...