Blackberry sip softphoneemplois

I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory

...Phone. The target uses ARM architecture, runs Pine64, an embedded platform and has Node.js based API running on Linux. Deliverables for the POC: We would like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsite APP communications. Establish a webRTC to SIP(PSTN) connection having two-way audio call and local VoIP connections (Guard Phone). If an alternate viable method is proposed we can consider it...

Hello, US accent agents wanted. Hiring worldwide for remote agent support. Must have a good internet connection and a softphone with phone number we can forward calls to. Please feel free to provide a link to a voice sample. To start you will log in to our answering service and return calls. You will have the opportunity to ask any questions about your response to the call/s prior to any call backs. Once you are trained and understand the service and are suitted with the conversations you will be able to take live calls. Base pay plus commission. We are a generous company and the services you will be answering questions to will be between 500-2500 US. Commission percentages will be based on sales and performance. We do not want you to have a "sales" personality....

Need help integrating DR3000 repeater with SiP based system

Hacer una práctica en Linux Debian en su versión 10.1.0, Asterisk 11.25.1, softphone Zoiper o Blink en sus distribuciones Free y configurar la Seguridad en SIP dentro de un mismo escenario (Máquina Virtual), Wireshark, VirtualBox

hello Desginer We'd like to design 2 banners for Collagen Sip Straw with following contents ( key message) 1. Radiant Beauty with Collagen Sip Straw from Japan for 4 weeks 2. First Collagen Straw in The world Reference: - - Thank you

Looking a freelancer who has experience in building International SIP ConnectionSIP system

I need to run the my voip server with Asterisk. I also want to run the x-lite softphone from the VPS as well. What I need: Set up Asterisk on my server. Configure my SIP account from my number provider to voip vps. Configure

We are looking for a skilled developer or team to develop a simple one click remote support for our team and custom...open source solutions or from scratch according to your preference as long as it includes the expected , I wish to mandate you to take care of my private server hosted at OVH in public cloud, I set up and installed the solution "GoAutodial" in v4, unfortunately I meet bugs with "Kamailio" on the telephony part in webrtc in the browser. I explain myself, everything works correctly, the SIP is well connected in asterisk and I can make automatic calls on the campaigns, unfortunately one call on two I do not hear sound for the interlocutor. I tried everything to debug it, I can't manage it. I don't think it's a big deal, I'll let y...

I have Jitsi configured on my server but facing some issues with installing Jigasi for Transcription and SIP. As of now, I need to enable Jigasi for Telugu language, so need to understand how we can configure multiple languages in Jigasi. Google speech to text account is free for 1 hr so, you should be able to test and configure using that.

Configuratiion Push Server to wake up our mobile SIP dialer to a pBX

press prefix 9 or 90 it should take sip line of yeastar S300 PBX and call for Outgoing

For a voip and sms provider based in Europe, we're looking to generate sales based on SIP-trunking of SMPP trunks for SMS. Connect with the VOIP providers of your country and introduce our company. Generate the contact and introduce ourselves.

...router/firewall donde quiero que funcione BigBlueButton. Como el operador de comunicaciones (O2) no me da IP fija accedo al exterior utilizando DDNS. Todo esto ya lo he ehcho y en principio funciona pero tengo un problema: Los usuarios externos se conectan pero la prueba de audio no funciona. He oído que es un problema de que los paquetes UDP no cruzan el servidor aunque he visto que el mensaje SIP que devuelve mi servidor va con la dirección IP local y creo que e spor eso por lo que no funciona. ...

...ideal for a morning water workout, to work on your tan while you sip your favourite beverage, or just people watch. As day turns to night, relax on the comfortable deck loungers with spectacular views of the city. Sip on a drink of your choice from our large selection of premium wines, beers, signature cocktails, join us poolside for the daily sun-downer cocktail ritual. - We call it "The Edge" because it was decorated with a transparent glass perched on the very edge of its own infinity pool which makes you feel like you swim to the horizon. USP 1. A tropical bar style & a transparent glass on the edge create a unique experience to the guests. 2. Variety of food & beverage selections. It’s the place you can have a sip on a drink of your choi...

Building a custom android app via ionic. adding linphone SDK in capacitor . UI layout is done just need to build out the core classes and connect the dots. and do SIP testing SIP testing Make sure SIP Call works 2 way audio Make sure SIP DTMF works (SIP-INFO) Make sure RTP is flowing correctly Make sure SIP Inbound CALL is working [do we need a push notification server] Make sure SIP Register is working with the correct Timers in place .

I am looking for someone to help me develop 5 of my tee shirt designs into graphic media png transparent format. I need someone with a creative and artistic eye for an ethnic clothing line. 1. Have a nice day. (help choose a font and edit smile on emoji). 2. Blackberry “SWEETER”. (help choose a font with color to match the berry and realistic berry picture). 3. Blackberry cluster (realistic berry picture). 4. MMXX (help choose a font). 5. I’m with Harriet (edit picture and choose font).

Requirements : -Build a VOIP portal with concurrent calling for use with SIP trunk providers.

I want to make my PBX outbound SIP calls to pick the next available SIM slot in cycling manner if the previous is busy, Setup SMS sending & receiving using SMS API, Call Credit Limit management, SMPP for SMS. Experienced freelancer with dinstar uc2000-vg can bid

Hello, Freelancers I want to deploy Android and iOS app using React Native. You must have experience with SIP JS. Please share your app using Thanks

What we need from you? Need a story or concept to promote the core idea mentioned below that can be expressed through Sand Art movie. Approximately 1000 to 1500 words to explain the idea. No google concepts or Youtube Plagiarism Only original ideas welcome. Core idea: We need to c...promote the core idea mentioned below that can be expressed through Sand Art movie. Approximately 1000 to 1500 words to explain the idea. No google concepts or Youtube Plagiarism Only original ideas welcome. Core idea: We need to communicate the concept of Mutual Fund to general audience in an innovative and different format than general Mutual Fund companies. You can use the below features: SIP Tax Savings Discipline and Patience Short term and Long term investments You can submit in PDF or Wo...

We want to develop an softphone app - VOIP phone The voip server will always be the same Example: Android & IoS

Hello, Freelancers I want to deploy Android and iOS app using React Native. You must have experience with SIP JS. Please share your app using Thanks

...Normal extensions won’t be able to dial external telephone numbers - Extensions are remote, so we will have to be able to configure those on Gigaset VOIP phones and SIP software phones as well - Extensions will usually be simple PBX extensions (i.e. 595) but they could occasionally be external numbers (in the last case scenario, we will need to keep control over the call, still recording it and being able to disconnect, whisper, listen and so on) - Extensions must be distributed between 3 queues with a skill based routing criteria where i can assign a level of skills to each extension - You will have to setup 2-5 SIP trunks associated to the queues This is a paid phone services business, so users pay in advance a variable amount of Euros for a relate...

This project is for development of HTML, CSS and JavaScript to implement the user interface for a Chrome extension for a VOIP softphone. Phone functions (making and receiving calls, etc) are not part of the development scope. Only the UI needs to be built; connecting to the actual phone system will be done separately. Javascript to be written in plain JS without frameworks such as React/Angular/etc, you must be an expert in DOM manipulation without use of frameworks or helper libraries. Design will be provided as an Adobe XD file, and we have a specification document for how the Javascript interface will work.

I need help getting the liblinphone sdk up and running. I’m a software developer, but I can’t seem to get the liblinphone sdk built. I’m wondering how much it would cost to get a tutorial from someone to help me get up and going. All I need is simple call functionality for inbound and outbound calls as well as being able to transfer calls to another sip account/phone number. Also, I need to be able to build on iOS, Android, macOS and a Windows.

...designer to join our team. Based in Los Angeles, CA, our brand is committed to providing fair trade, ethically sourced and environmentally friendly ingredients from around the world. With a variety of offerings designed to calm your mind, relax your body and soothe your soul, each botanical blend is infused with organic water-soluble hemp to ensure a consistent and calming experience with every sip. Job Description: Freelance Graphic Designer will combine their creativity and technical skills to consistently create digital graphics and images for website, social media and marketing use. We are looking for a graphic designer who’s passionate, timely and experienced in e-commerce to help elevate our brand identity. Key Responsibilities: Work directly with company founder and...

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

we need to get assistance for app-build on ionic-cordova-sip getting this to work on SIP. using that github (since it runs with cordova/ ionic / angularJS) our SIP server does not support webRTC so we have to go native for this to function. not sure if MWI works in liblinphone stack (or can we code this to accept that SIP method notify. we might need also consulting time for SIP setup VoIP expert in this kind of level with appDev skillset as well .

We're a small business that has multiple development projects ongoing. We're an Atlanta based company and are looking for a local developer that we can work with. We're in need of a Full Stack Developer that has experience in: • LMNP (Linux, Mysql, Nginx, PHP) • Redis • Laravel • REST APIs • Vue.js • Node.js • Express • MongoDB • • Docker Pluses: • Asterisk • VoIP • SIP • Networking • Linux

This project is for development of HTML, CSS and JavaScript to implement the user interface for a Chrome extension for a VOIP softphone. Phone functions (making and receiving calls, etc) are not part of the development scope. Only the UI needs to be built; connecting to the actual phone system will be done separately. Javascript to be written in plain JS without frameworks such as React/Angular/etc, you must be an expert in DOM manipulation without use of frameworks or helper libraries. Design will be provided as an Adobe XD file, and we have a specification document for how the Javascript interface will work. Please show me your most similar project you have developed bidding

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

FREELANCE ARTIST WANTED! TAFE NSW Event students are holding a livestream paint &sip in partnership with Youth Off The Street. We are looking for a painter who is comfortable leading a class online, and can supply their own materials. (excluding canvas) You must be available on November 4th from 4pm onwards. You will be on location with a set design and layout for your convenience. Amazing opportunity to reach a large audience across Sydney and be a part of a charitable event. *We hope you can offer your services out of goodwill. Otherwise, a sponsorship package can be arranged.

iam looking for an android app which is similair to silver dialer/i tel dialer (referrence ) which is compatible with g729 audio codec & works with all standard soft switches. works well with minimal bandwidth

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

The app is 2 pages (+ login page) Pag...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

We are looking fo...separated in two parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show bal...

We are trying to build asterisk system based on the following requirement. Asterisk (R) and Asterisk (B) Asterisk (R) which is Raspberry PI based and attached with USB dongle with no SIM card Asterisk (B) is connected to some SIM box but with SIM card attached Using softphone, we are connected to Asterisk (R) or Asterisk (B) and make a call to some number. Asterisk (B) will send some code to Asterisk (R) to remote link up the cell / mobile signal via IP to initiated the call Both Asterisk (R) and (B) are ready, and your job is to link up the (R) and (B) so (B) can link up cell / mobile signal remotely

... up and working and proxying to SIP server. i have built out but i might be missing something so small need someone to help troubleshoot to get it working.. i have 1.) use or hopefully to verify webrtc client demo (login with sip:1111 @ with wss:// ) SIP PBX <---->Kamailio <----> webrtc client () the config im using is this basic need = i need to use webrtc to register as sip and make a call like sip (zoiper on the web) after research i need to do this with WSS:// since our PBX is only SIP this is the route im going so im trying

Hello, would you be able to help us provisionning some equipment (SIP phones and routers) on GenieACS 1.2 ?

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/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gateway ...

Need a business logo to represent a fun creative art class studio. Where you can sip a beverage while being creative painting Business Name: Harmony Art Tagline with Business Name: Sip, Paint, Create

Need a business logo for an art studio offering sip n paint fun art classes. 1. 1st logo to just read: Harmony Arr 2. 2nd logo as above but slso incclude tagline Sip n Paint

i want to convert my android phone as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. (VOIP CALL TERMINATION) so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723

I need an Calling Android app. for VOIP Business like Itel Dialer with features of register with email and recharge or with Pin Sip by company

mini POC WEBRTC-to-SIP architecture backend buildout for SIP server to webrtc (kamailio or janus) docker build setup to Sample this SIP-WEBRTC (to enable websockets) client= webrtc sip register=SIP server does RTP have to flow through kamailio or SIP-WEBRTC as well trying to model off this

Need a call to provide a challenge on Twilio sip trunking solution.

hello, we would like to create a web softphone for our asterisk freepbx server. the page must have username and password to access the web phone. on the same page there must be information such as, campaigns enabled, calls waiting, calls answered, and average conversation time. all on one page. then we would like to have a control panel where to enable users to the various campaigns / queues

Hi, I need you to configure opensips on my ubuntu server, it has to support calls from linphone to linphone, sip messages and new users creation.

we would like to have both the sip client and the telephone bar where the user sees the code where it is inserted and where there is the possibility to pause the sip