Logiciel sip blackberry jobs

...chasing a bottle or glass of whiskey. Both should have legs and look like they are running. Must be suitable for screen printing (or similar) on shirts, so heavier line weights are preferred. No cartoon faces, please. The line drawing or illustration should be black and white as it will be used for a single-color screen printing. Meant to represent a "pickleback" where each sip of whiskey or rye is followed (aka "chased") by a sip of pickle juice - sounds gross, but it is surprisingly tasty. Why? Because it is the birthday of a friend. He introduced a group of us to drinking picklebacks AND we all run together - so we want to silkscreen a few "team pickleback" running shirts for the group (he will get his free, of course, as his birthday pre...

1 Kamaillo is already installed, I need Kamaillo configured as a SIP routing server/SBC and data replication both Kamaillo 2. Data replication between both Issabel At the end of the project, I should be able to. , Register sip trunk at the Kamaillo . Register a phone at Kamaillo . Make and receive calls. . Data replication between Issabel PBX . Data replication between Kamaillo SBC

The app is 2 pages (+ login page) Page one: Dial pad showing the credit and the cost per min and remaining minutes + control the call when the user exceed the limit page two: Option for the users to buy credit ( in app purchase ) Spin numbe...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 d...

The app is 2 pages (+ login page) Page one: Dial pad showing the credit and the cost per min and remaining minutes + control the call when the user exceed the limit page two: Option for the users to buy credit ( in app purchase ) Spin number...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 d...

We need a React Native sample app that connects to a sip server and makes audio call. It needs to be like a phone call, using ear speaker proximity sensor etc. Preferably jssip, react-native-webrtc, react-native-callkeep libraries should be used. Interface is not important, we will integrate this sample code to our app.

3CX Sip Integration with CUCM and Alestix

3CX Sip integration with CUCM and alestix PBX

...to develop a SIP to Viber/Bip apps gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/Bip 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 Viber/bip 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 Viber/bip gateway 2) Viber/whatsapp ...

...to develop a SIP to Viber/Bip apps gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/Bip 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 Viber/bip 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 Viber/bip gateway 2) Viber/whatsapp ...

Design, specification and working drawings required for pre-fabricated studio/home office/ school classroom, will also require full specification and working drawings. The structure will be constructed using a timber clad SIP system, i have up loaded similar buildings for guidance, but we would require a unique design.

Looking for someone with skills in: ReactJS Redux PostGresSQL (Sequelize) Pro in Asterisk and WebRTC Excellent Web responsive developer COMMUNICATION IS IMPORTANT TO ME, GOOD, RELIABLE COMMUNICATION - !IMPORTANT WEBRTC WITH SIP via SIPJS, JANUS, or SIPML5 - !IMPORTANT SKILLED AT MAKING DIALPLANS IN ASTERISK AMI AND ASTERISK AGI - !IMPORTANT NEED SOMEONE WITH VERY GOOD ENGLISH - !IMPORTANT The application is in the process of being built. I need someone to take care of the additional tasks. I need someone who can dedicate their time and provide clean and organized code. 1. "Build in call routing into the groups. Inside the SMS Group settings, when a call comes in we need to be able to decide if the call will: - Be forwarded to ring the last agent that spoke to the customer w...

CRM company looking to add a multi-tenant PBX solution. Currently using Asterisk but looking for something better to handle many simultaneous customers and to API with current software (php, mysql, ajax). Require the ability to add, edit and remove SIP users along with codecs, voicemail and call history. Will use Twilio for origination and multiple wholesale providers for outbound. The engineer awarded must create the AWS linux instance, configure PBX and assist in providing API information. Additional maintenance work will be needed.

calculate load from one SIP wall with external finish. full structure design including beam has already been done. Just one sip wall load needed.

SIp client for presence, registration, chat, audio call, video call, and group audio/video call.

We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server

Kiosk mode, SIP softphone in C or Java for linux as per the pdf attached Note: any paid/third party component would not be acceptable.

I want to create a sofphone for Windows using Linphone SDK Softphone should be able to dial direct from GUI and I should also be able to send command from other application to dial X phone number Program must be nice GUI and I will need source core with all libraries once paid in full.

Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 7. Server A to Server B voice traffic will be e...

Asterisk 17.4 is downloaded and extracted, but needs to be properly installed onto a server for us. PJSip config to a SIP provider, the ability to provider music/advertising on-hold, a couple simple queues. We do not want freepbx, please do not suggest it. Just install asterisk and, as needed, help get config files setup.

We are a telecom startup (with existing customers in beta and ready to launch soon) and we’re looking for a PHP full stack web developers with experience in twilio and having some knowledge of telecom, not 100% required but would also be great if you have knowledge with FreeSWITCH & OpenSIPs (both open-source telecom software & PBX / SIP Server). You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line Ubuntu/Linux/SSH MySQL GIT Twilio These skills are optional: AWS Facebook APIs React Telecom knowledge FreeSWITCH OpenSIPs We will offer long term work every week but only if you prove your skills in the first week. We are only looking for serious developers and to prove that please fill the attached docum...

We’re developing a business to build and sell back yard offices in the US. The idea is to build three variants: 64, 96 and 112 sq ft. The materials specifications we’re looking for: - Walls, floor and roof structure: 4” PU SIP panels. - Double glazed timber door system, fixed side lights and top swing window. - Roof: single ply flat roofing membrane. - Cladding: vertical red cedar cladding left to weather naturally. - Vinyl flooring. - MDF painted skirting. - Walls fully plaster boarded. - Lights: 4 chrome LED lights inside. 3 outside. - RCD breaker. - PVC gutters and downpipes to rear of pod. - Some illustrative pictures are attached. What we need is a professional with experience in the construction business to help us with: (1) Validate specifications with ou...

Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

Install Asterisk in the same 3cx Server make a General SIP trunk with DID number from one of your internal extensions, then add to this trunk DIDs for all of your internal extensions; You need to add inbound rule for each of the trunk DID and configure it to route calls to corresponding extension number. Integrate on Vtiger and zoho CRM , noting that Vtiger is on cloud server and thus the Vtiger and Asterisk Server are different location, then MySql need to be open to connect Vtiger and Asterisk.

Project for MD Implement SIP for voip on matrix server.

I want to implement sip phone into our application using Delphi Language with Linphone SDK But since the the DLLs are in C++ library, we need the header files to translate the C function header to ObjectPascal (Delphi) calls so we can use it in Delphi.

I registered for an online sip provider. When I use their app, I can add caller-id's by verifying phone numbers by sms. Without verification, changing to any custom phone number, is not possible. When I use the same SIP from my own asterisk server. I am able to change the parameter CALLERID(num) to ANY value, and without verification, the caller ID shows up on calls. I would like to use the provider directly, without integrating it into my own asterisk server. I have an open source iOS app = linphone. I can set custom Headers etc.., but I have not found a way to set a custom unverified number as the caller ID. Objective: Provide information which will allow me to call from any caller ID using the open source linphone swift iOS app.

This is an online talk show with 6 women discussing current events, relationship issues, etc. Sort of like a Zoom session. The show name is The Ladies Room Live! We are looking for a hot logo.

Basic Website is required to have following features: Display all my products [INVESTMENT | INSURANCE | REAL ESTATE | LOANS | TAXATION]. (Just basic information required to be displayed.. . One page for all product information and second pag...TAXATION]. (Just basic information required to be displayed.. . One page for all product information and second page for Real Estate advertisements) One database should be connected to maintain customer/ lead provider details. Customer should be able to view their investment details in respective logins. Premium reminder to be sent on customer's mail/whatsapp Few useful tools [calculators] [i.e. SIP/ SWP returns. Loan calc etc..] You can take it from existing website. Note: We can discuss it in detail before finalizing My Budget is ...

I am looking to buy a dialer or a web application that allows to send traffic missed calls with the feeling of adding a new sip and caller id and the incorporation of contact file

I have installed openSIPS server but need to configure it so that it does the necessary basic functions of SIP server with modules such as the SMPP for sms activated ( & ).

The company is called ‘cool sip’ they will be providing stainless steel and silicone baby bottles with zero plastic. But will also have a range for adults in all different funky colours and designs. They will keep Cold drinks cold and warm drinks warm For hours. I want a simple but ‘cool’ logo.. (Similar time the chilly logo for the company ‘chilly bottles’) I want 2 things a simple Cut cool logo that when people see it they will know exactly what company it is And also the brand name design (cool sip) which will be engraved on the bottle and packaging. Want it to be cool and stylish but also stand out. Good luck

...button) On the screen: -Textbox: user can enter the username; placeholder: Enter Name -Text: “Please enter your name or organisation only! Do not use this app for any harmful purposes. We record all calls and make them accessible for you or the relevant authorities on request.” -Save button -Haptic feedback if username was successfully (not) saved -After save send the username as SIP name to the SIP server ...

Build audio and video conferencing and podcast site with webrtc and sip using BigBlueButton(BBB) Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .I will provide mockupsbof new layout and design for you to are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

Hi Sohaibur R., I noticed your profile and would like to offer you my project. We can discuss any details over chat. i need configurate whatsapp like sip trunk and make current calls

Hi Mohammad Waqas Shoukat Ali S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. i need configurate whatsapp like trunk sip and make current calls

We search an Android/IOS and C++ developer for Windows app. Require modification of an OpenSource application with its own functionality and remove some functionality from the frontend. The documentation of every modification is required also. Skills Required: -SIP -VoIP -AT LEAST 1 of the following: C++ OR Andriod OR iOS developer PRICE AFTER DISCUSSION, just place your offer

Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

Hi, I want to to use any mobile phone as sip gsm gateway

Hello: We need to fix an issue with our Big-Couch Kazoo DB for Kamailio-freeswitch system. I think the DB system needs a better node configuration. Will love to work with a dedicated passionate professional.

I need a system of operators who can speak only by voice via webrtc. Whereas I need a register of agents that the call can arrive with anyone who accesses the channel. If integration of SIP channels with WEBRTC is possible, it is desirable to

We have an Entry Phone App which we need to improve. We wish to expand out the number of features offered to user of the app. The current interface is too limited and cumbersome to offer our proposed feature set. We wish to develop a smartphone APP in both the iPhone and Android platforms. This is a large proje...Android platforms. This is a large project, but for now, we're only looking for a freelancer who can complete the first stage of the project, which is to include a Video Call feature on it. The tast: A simple APP on the Android and iPhone to demonstrate your firm’s competency to create a two-way audio/video call from the smartphone to our target Entry Phone. Preferably, we would like to see a SIP VOIP call placed through our Asterisks Sever. We are open to a...

I need to developer who is good at SIP AND SMS. A standard SIP client running on the separate android phone will have to connect to the SIP server that was developed on the other android phone via Wi-Fi and an incoming and outgoing call would have to be demonstrated. A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. For example, using cellular networks to send automated calls and messages to your phone contacts. The current Android phone is unable to do that. who is can provide call and media control that is required to make them work as a gateway? Please add "I know" to the first sentence of the bid. thank you

I need to developer who is good at SIP AND SMS. A standard SIP client running on the separate android phone will have to connect to the SIP server that was developed on the other android phone via Wi-Fi and an incoming and outgoing call would have to be demonstrated. A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Please add "I know" to the first sentence of the bid. thank you For example, using cellular networks to send automated calls and messages to your phone contacts. The current Android phone is unable to do that. who is can provide call and media control that is required to make them work as a gateway?

I'm looking for someone to make a user application for discord that bridges a SIP audio stream with the discord voice channel -SIP to Discord audio stream

i have issue with my call center

Build audio and video conferencing site with webrtc and sip.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

Need to install and configure the Astrix Server, for voice calling. Below are the resource. 1. Cisco IPPhones 2. Having white list IP and voice port, provided by ISP 3. Don't have sip trunk and DID number. Is there any possibility to make outgoing and incoming calls with these resources ?

want to config SIP IP Phone Setting within my Local and internet setting Pictures are Attached hardware phone. i Have three phone sets and want use at lan network first and then within internet. Hardware is China Base and Language is also china. Want to change Language too. web address of

...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 gatew...