Sip app blackberryKazi

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

I am looking for a team or duo, who will develop an app for iOS and Android frontend/UI. We will develop the backend. Android and iOS app based on the open source project: Linphone () Download the Linphone app to your mobile to see what we mean by to do. (You will have to modify that open source project) To do: Deactivate top menu line Deactivate slide-in menu (keep it but make it unaccessible for users) Deactivate chat (bottom menu) Deactivate top menu of Contacts Deactivate ‘call-recording’ on dialer screen on the button “1” Deactivate save number button on dialer screen (Deactivate = Make function vanish on UI as it never existed = unaccessible for user, but keep the function in the code: no need to waste time by removing code) Contacts m...

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 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 alternative metho...

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...

...consumers from the WhatsApp app on their smartphone. We need to develop a SIP to Whatsapp voice gateway to send some of the calls from our PBX. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's WhatsApp number. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to t...

We want require Voip Developer good Knowlage our ASTRIK SIP Calling Coding work at leat 3 year min. Development of SIP Calling Work

Sip-n-Cycle Pedal Cruise is a tour boat on the Alabama River in Montgomery, Alabama. This is a fun, happy, colorful flyer for tourist and locals to learn and Book Now on . The attached pictures are NOT MY BOAT and so when you see Paddle Pub, that is not my boat name. My name is Sip-n-Cycle Pedal Cruise. The background of the boat is not like the area I live. The company building the boat has given me pictures to use to promote my boat coming to downtown Montgomery. So if you use the Paddle Pub picture, blur out Paddle Pub and/or put my logo over it. Comparable fun, sample brochure attached. Book Now at Phone number - 334-399-2387 Bachelor/bachelorette parties, celebrating birthdays, family reunions, corporate team building

Hi there, We are a telecommunications company based in Australia, with a modern, cloud-driven voice infastructure. We require a simple SIP stack built in AWS, no RTP required. This stack should be built to receive a SIP OPTIONS message and respond with a "200 OK" including a custom header message. We wish for the value of this custom header response to be pulled from a NoSQL database on the fly - e.g. redis, DynamoDB, etc. The method of determining the value is up to the author - e.g. the source IP, byte length of packet, etc. The conditions of successful completion are as follows: * The stack must be built in AWS * The deliverable must be on GitHub (public or private and shared, either is fine) * The deliverable must be deployable via CloudFormation * The infra...

We have a running jitsi meet server, but we have some issues left to complete customizing We ask for the following issues: 1. multi-domain support with different logos and imprint and GDPR statements 2. sip connection with jigasi 3. coturn server configuration on different subdomain 4. configuring ufw firewall 5. optimize jitsi performance for more participants 6. authorization for opening rooms 7. configuration, if recordings are allowed for the moderator 8. if possible muting from the moderator How long do you plan for this project and how is the payment you need for the project?

Hi Amal, I need to install asterisk server and setup intercoms between 100 extensions. There will be sip trunk which I need to configure so that outgoing call will go from that trunk. There are few more details I would like to understand like feasibility of the solution. Let's connect to understand it further.

We want to create one honeypot project where we want to create fake service as below ystem Objective The objective of the proposed system is to develop a HoneyPot security System. The HoneyPot system w...3. Web(HTTP) Captures Login attempts. For this purposes a dummy login page will be presented to the attacker 4. Telnet Captures Login attempts 5. Port Scan Captures Port Number attempted to be accessed 6. Windows Shared Files (Samba Protocol) Captures File attempted to be accessed via Samba 5. MS SQL Captures Login attempts 6. MYSQL Captures Login attempts 7. SNMP Captures Old Requests 8. SIP Captures SIP requests 9. VNC Captures Login attempts 10. REDIS Captures actions 11. TFTP Captures requests 12. NTP Captures NTP requests 13. TCP Banner Captures connection and subsequ...

SIP configuration for inbound calls of vici server

We're looking for assistance in adding UUI Headers to a SIP Call, ideally using cloud-based or on-premise telephony equipment. For instance, some equipment to serve as proxy between the source and destination. We're looking to transfer a call from a system that doesn't support SIP headers currently.

...Debian Linux, which I installed myself and verified that it works. I'm connecting it to Plivo's Zentrunk. Outbound trunking works fine. But inbound trunking doesn't. When I allow both *Anonymous Inbound* and *SIP Guests*, it works but I'm flooded with spam calls too. When I disable either one, it quits working. Plivo's inbound trunk origination URI has authentication entries for username & password, but I can't seem to link that with FreePBX. "Allow SIP Guests = NO" shows this in the log: [2020-04-30 06:02:08] NOTICE[18715][C-0000125f] chan_sip.c: Failed to authenticate device "Me" <sip:+>;tag=gK00139f73 That's my real phone number and CID, and it shouldn't be authenticating that specific ph...

Initial Preferred CMS: Wordpress for CMS, articles and blog Phased development approach: Phase 1 - simple informational page(s) Top menu bar - menu pull downs - corporate logo, client ...and help desk login Banner - rotating - clickable (graphic image, security) Sections - services, blog entries Partners - scrolling? Customers - scrolling? Bottom bar - standard stuff, addresses, contact us form, social media, blog. Example sites: Phase 2: Integrating services, SaaS and products, SaaS front end This phase will be contingent on skills to work with .net, angularJS, React etc, in building dynamic front ends and dashboards.

We are looking for an inside sales person to follow up on leads and close sales over knowledge of Telephony (SIP trunks) would be an advantage.

We are looking for our own Softphone sip Dialer.

...project is to create a responsive video conferencing website for multiple users with frontend in Vue JS and backend in Node JS with WebRTC hosted on AWS. Get in touch for further details. Skills Required: HTML, JAVASCRIPT, JQUERY, CSS, JSX, VUE.JS, SQL, AJAX, AXIOS, NODE JS, AWS EC2, WebRTC Experience with WebRTC platform including the SIP, RTP stack & SDP, RTCP, TCP, UDP, SIP, HTTPS, SSL/TLS protocols. Experience in VoIP products . Integration of WebRTC to SIP using Jitsi for Web and Mobile Applications. Knowledge of WebRTC server Strong proficiency with GIT, Node.js, JavaScript/ES6 Experience in developing center products and solutions & integrating third-party or open-source solutions. Strong competencies in data structures, algorithms, and software de...

The objective of the proposed system is to develop a HoneyPot security System. The HoneyPot system will be a Low interaction system which will be installed on a system and will then capture attempts to ...3. Web(HTTP) Captures Login attempts. For this purposes a dummy login page will be presented to the attacker 4. Telnet Captures Login attempts 5. Port Scan Captures Port Number attempted to be accessed 6. Windows Shared Files (Samba Protocol) Captures File attempted to be accessed via Samba 5. MS SQL Captures Login attempts 6. MYSQL Captures Login attempts 7. SNMP Captures Old Requests 8. SIP Captures SIP requests 9. VNC Captures Login attempts 10. REDIS Captures actions 11. TFTP Captures requests 12. NTP Captures NTP requests 13. TCP Banner Captures connection and subsequ...