We're looking for someone who has previously written modules for Asterisk. Modules that preferably were constructed to work in version 11 and 13. The ideal candidate will be able to stand up their own Asterisk development environment to build and test in. An asterisk module which will allow us to, while placing an outbound call and playing a set
...are building a VOIP project which requires SIP Trunking methodology and integrating SIP Trunk with Asterisk server. (Asterisk version is important, while you posting at which version can you able to deploy) - Need to have excellent skill set in Asterisk, PBX, Integration of SIP Trunking - I must able to make calls thru Asterisk. I will ...
need help fixing a vicidial server when the agent logs in the initial call doesn't come over for 45-60 seconds well after the interface times out
...design that is similar. All the pricing and details can be found on the website: [kirjaudu nähdäksesi URL:n] On the flyer I need date, time, location, ticket pricing (with an asterisk that the prices are early bird pricing) I also need room to add Sponsor Logos which I will add later. The website must be listed as well. The tickets should have all that
...(depending on the changes), you only have to make the integration with the API and of course you will have the documentation of all services; that means that you don't have to write code for logic (business rule), the only logic that you have to build is the UI of the mobile app. Our development team will work in the logic, so, you will have to call the HTTP
Necesito configurar una centralita Asterisk para 5 puestos de trabajo y darle acceso al exterior a través de la red de fibra contratada, en Madrid. Busco un profesional en instalación de este tipo de centralitas en oficinas y que pueda realizar el trabajo en Madrid.
I need a new website. I already have a design, I just need you to build my online store. please see the design and the bid link: [kirjaudu nähdäksesi URL:n]
I need a UI creative designer to use his/her creativity in building the initial design for an eCommerce responsive website in both RTL and LTR direc...variations. - Materials would be very similar to this site (freelancer.com). - The idea will be provided in details on chat. Important: please add your total price for the Bid (not per page). Regards,
... Profile picture upload issue 2. First Name and Last Name in 2 separate boxes should be instead of full name. 3. Fields validation issue (marked by red asterisk) 4. Target Job location (the list should include drop-down menu with Qatar cities and zones, multiple choices option should be available) 5. Job Industry, Career
My Project Scope Is Coding Asterisk Virtual Server. 1: Code Asterisk Server And Configure To Be Used In A Local Area Network With A. Hard Phones B. Soft phones Application Installed In Android Phones IPhone Phones N:B The Coder Will Recommend The Hardwares A. Asterisk
...Maintain and manage Genesys Routing, Framework and reporting. Responsible for supporting call center routing strategies and have experience with Genesys reporting, URS routing, SIP and GVP technology. Good understanding of Genesys (CTI) infrastructure. Strong working knowledge of IRD and CME. A working knowledge of Genesys Systems Architecture to create
Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee
Please read requirement before bidding so when you bid The price dont go higher, I need an iPhone/iPad and android app with website . I would like it designed and built. Food network Review Foodspy Multi language arabic english -showing restaurents (stars, category, and with ( recently added , recommended, related to your views, sponsored) - allow
I have a production asterisk installation running on my server. I have a requirement. I want to setup a queue such that Agents and end users can use queue using their mobile phones. Lets Say, their are 3 agents Agent 1: Mobile : +91-XXXXXXXXX1 Agent 2: Mobile : +91-XXXXXXXXX2 Agent 3: Mobile : +91-XXXXXXXXX3 Lets there are 5 users who will dial
We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone
Hi i'm looking for someone to do a video logo for my son's 1st birthday party. check the attached image for sample Logo, the text should b...stay still, background should be bokeh light effect. just moving, text will not move, just background light will glow. PLEASE read it carefully and check the photo before you bid. Need it before tomorrow morning.
I am looking for someone who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the re...who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the required graphics and we also have an internet facing SIP server it will register to.
...(inclusive) and store these into an array. Produce a chart EXACTLY like the one below that indicates how many values fell in the range 1 to 10, 11 to 20, and so on. Print one asterisk for each value entered. Notice the spacing for everything. Range # Found Chart --------- ---------- -------------------------------------------
Hi altr, I'm looking for someone to help me r...using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure this SIP message is forwarded to the customer
I have put multiple projects of like this and have not hired no one, the reason is I am tired of the same responses from the same people, they are all very generic and made for everyone. I want something made for not just me but for my company. I need S.E.O done for a web development company. I want a specific analysis and plan for my company not a basic one that is given to every company. M...
...ou seja faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar out...
...'moodle'): [kirjaudu nähdäksesi URL:n] Required fields are next, where are same fields obtained from the original report, plus (marked with asterisk*) four fields that will be calculated using the same extracted data: - Date and time - First name / SurnameSort - Email - Grade item - Original grade - Revised grade -
We're looking for a senior React Native/Redux developer to complete the final steps in a VoIP/Text Messaging mobile app. The mobile app uses PJSip to communicate with Asterisk, and interacts with a back end API created in PHP. Ideal candidate would have knowledge of React Native, Redux, and PJSip (Optional but definitely recommended). The final stages
...script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations with the top officials of the companies; Sale of services in the field of B2B; Requirements:
play m3u8 stream inside stream display a button call ( ...without stop playing stream please don't waste my time. if "Bid ok then no ask for more money after" or "want deliver money before start to work" or "be sure you re able to do the apk function call speaker on" or "need 45 days" etc .. my budget is 200 US$ read it carefully BEFORE BI...
Hi , I am looking for someone who is able to configure ZRTP on freepbx ? and would like to know if all features work with this protocol? I read some articles said that call recording is not possible with ZRTP. Does it work with all other protocols such as, TLS, UDP and TCP? Let me know if you are interested in this kind of work and let`s discuss the duration and price.
We want a site where we easily can see the calls that come in and what happens to them. An example could be A customer calls 70209404 (NordicCall), the customer is in the queue for 2 minutes because every agent apart from two are on DND, one agent is busy an the other agent rejects the call. So we must continuously be able to see all of the information, and there is a site with further informati...
...and, if they are valid in system. If so the backend server would send call set up info (dial plan) to asterisk server (I have an asterisk Guy to work with). The app would then dial an 800 number that was returned to the app on call setup. Asterisk would match the customers CID and process the call as per the ad hock dial plan created for that call substituting
Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload
***Note Please only bid with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain
Hello, 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/Android, or by using the
...persons and each one will need to have his unique click to call button. The registered client will after buying credits (already working) will choose the tech person and will ask to be called to his phone number (phone number, not webrtc). Before the call, the client will choose the call duration. He can be billed 10€ for 10 minutes or 20€ for 20 minutes
Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gat...connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls
...expert. Project: Telephone Answering Service Software Project Type: Software + API Integration Software Name: Switchboard Timeline: 2-3 weeks Phase: #1 User Types: Admin - login access Agent - login access Customer - login access Admin - Features Add, edit, delete Customer accounts Assign customer phone numbers (integrated with Asterisk ami
...back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers
Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.
...from iso on a bluehost server. I can register my sip phone, I can make inbound and outbound calls. The only problem is there is one-way audio on the call. If I call from my Windows Sip phone to my cell phone -- My cell can hear me, but I cannot hear the cell. If the cell phone calls my Windows Sip phone -- The cell can hear me, but I cannot hear