Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat and Facetime to complete the call to the called party number. The development platform/operating system is not important. The project should be completed
Linux Otra o no estoy seguro I need to install some software in a VM in AWS so the software can make a SIP videoCall to a Polycom device and stream the video call. I have been looking at multiple softwares like flashphoner, TrueConf or Pexip. The first step would be to select the best software for the job.
Microsoft access program with azure hosting. The access program has been configured into an auto dialer with a custom sip phone. The backend is hosted on Microsoft azure. There are dome features i would like to add, fix some issues, and copy the hosting to a new account.
Hi, Looking to hire open source contributors for interviews. The participation in the interview will get you $25. The interview shall take around one hour and various subject related to open source will be discussed. Example of questions: 1. How submit a patch work in your community? 2. How defects are managed? etc. There are no deliverables
...features as part of our package: Hosted PBX Call Recording Call Monitoring Real Time centre reporting (call costs, current spend, agent dials etc...) Voice-mail to email SIP Trunk Support (Remote and customer service) Number porting. Skills: Leads, Marketing, Research, Sales, Telemarketing See more: VOIP lead generation, we need sales agent,
...is like this : IP 192.168.0.111.sip > 192.168.0.110.avt-profile-1: UDP, length 533 E`.1....@......o...n.......SIP/2.0 180 Ringing Via: SIP/2.0/UDP [login to view URL];branch=z9hG4bK18438941-9cbe-4ee8-b4f9-68f611bf3677;received=[login to view URL];rport=5004 From: "8111"<sip:8111@[login to view URL]>;tag=kencgqon To: "8112"<sip...
I want someone who can make me register more than 1 user to receive call from my voip switch and send it to viber. i gave more than 300 viber accounts and i want to use them for termination. Project will get the developer good earnings despite the cost of the software or asterix.
Hello, Freelancers. This is the first micro project from us. We need to do modification for a code like subway surfer in this code we have 3character to change the environement and gui .
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements
...I currently use Natpass, but am looking for a more open solution. You must have built out this configuration in the past. I do not want to pay you to learn. - SBC handles SIP Proxy - Registrations - Trunking - Security by locking down only domains we add I will need to see what experience you have before I hire you. I would like also to do an ongoing
I am looking for someone, who can help me setup auto or a predictive dialer. Also, suggest a SIP partner.
...multiple options of server recreation and have deemed ASP or a C# Executable server to be the way to go. The game has to be modified in kind of a strange way, as we don't have source code for the game itself. We only have the SWF files associated to the game (see screen shots below). We've been using a tool called FFDec aka JPEXS Flash Decompiler to edit
...where my SIP for external calling is configured now i need a configuration from Goautodial server to my external CRM. So I want sip trunk between GoAutodial and External crm so my CRM is able to make and receive calls using Goautodial and use line that is configured in GoAutodial ; Because my provider only gives me 1 node to connect the SIP so it will
...looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work fo...
Need Mikrotik Expert Now to assist with SIP Audio QoS Issue - RB2011UIAS-2HND-IN. Some minor cut out of audio. Looking for someone who can start now, access firewall via my Teamviewer. Thanks, Gavin
...experienced with commercial setup and security hardening of VOIP servers. Below are details of setup: Project Description 1. Install Scalable Voip Platform Including Asterisk, a Sip Router a Billing System. 2. The installation will be on 2 VPS. Options include (Linode, Amazon EC, Digital Ocean, etc.) 3. Users can select and pay for DIDs online. DID selection
The goal to create with Asterisk video/audio conference room and connect it with jitsi clients. The Asterisk is installed on server. For the conference is conbridge module used. Jitsi should be installed on Raspberry Pi 3b+. After starting Raspberry Pi it should automatically dial the the conference room on Asterisk.
I bought source code from sellmyapp and I want add my ids for ads and in app purchase and Facebook SDK add my id for admob and chartboost and unity ads,add Facebook analytics achieved level event,add ids of products to my account in Google play , build source code to Android iOS Facebook webgl platforms
...the it: • Screen (either VGA or HDMI) • Wireless Keyboard and Mouse (USB Port) • Headphones (USB Port) • Power What must be installed on the machine? • Browser • SIP Client Who will use the machine? • The machine will be used by a call centre agent. • The call centre agent will open the browser. You must ensure that the browse...
SIP CALCULATOR: [login to view URL] HUMAN LIFE CALCULATOR [login to view URL] WEDDING PLAN CALCULATOR [login to view URL] CHILD EDUCATION
...user login by sip server user 2. show balance display 3. show call log 4. support incoming & Outgoing Call 5. Working Smooth 6. support g729,g729u & others voice code 7. Include any VPN Technology for SIP Block Country 8. Smart Graphical Interface 9. Include AdMob Ads **** When Complete Project then need provide full source code & full Documentation
...freelacer or company who can build a mobile sip softphone for us. that softphone must have all call features and locally store call details on device like dial/missed/received call. Softphone must be able to support audio and vedio calls with suitable codec. G729 is must for this. Users will be able to add multiple SIP accounts. if you already developed
I am making a data system. I have all of the code and arduino files, I just nee...directories to put them in so they can communicate with one another, installing and importing older versions). I need to use PyQt4, pyserial, and pyqtgraph. But for pyqtgraph I need sip, and am having trouble with that. I am working on a Mac, running macOS Sierra10.12.6.
Hi We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/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 Viber/whatsapp
...ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple callers id which to be round-robinly used for outgoing calls, if caller-id is defined for the sip-account then will only send calls
Few things that you need to do SIP applications are basically those applications using which you can call to mobile numbers through a network You need to do a little research about SIP calling You can find lots of free SIP calling credential from Zoiper for testing Below are the understanding of work for which you need to provide estimations...
ONLY Knowledge with VOIP bid. SIP applications are basically those applications using which you can call to mobile numbers through a network You need to do a little research about SIP calling You can find lots of free SIP calling credential from Zoiper for testing Below are the understanding of work for which you need to provide estimations...
Hello, i am looking who can change my source code, but do same things. APK file getting reject from google play store. - "violating policy repetitive content”. It's spam section. All source code is from codecanyon. Maybe need change it and add some new features prevent from google play banning. Looking who can resolve this problem. Please write
...will be 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
...Digital transformation and business strategy • Case study on a customer: I can provide content for you to structure an article with • Fall of ISDN in Australia and the rise of SIP Audio conferencing: • How to stay focused during company meetings (one point will need to be: using a high-quality audio conferencing unit) I need someone who has
Hello Everyone, I am looking for Akka framework, Akka persistance, event sourcing, CQRS and Java. Develop code using design documentation.
I need a plug-and-play WordPress website for a real estate management company. Site will need: 1. User portal with secured log-in for members 2. Back-end database tables (mostly accounting information) 3. Permissions on the database tables for specific users 4. Display member data in tables and charts 5. Needs to scale (100's of users) Site will live on Google Cloud Platform.
...Asterisk must not bridge the final connected call. Asterisk must remove itself from the call-flow once the call has been transferred. The other pbx is currently setup under SIP and PJSIP with the same remote ip address. We are using the Asterisk pbx as an auto-attendant only. Our main pbx will send calls to the asterisk pbx and expects the calls
Need someone who has a good understanding of SIP architecture. Task to be done: 1. Work on outlining the architecture documents. 2. Setup SIP server like flexisip, Astrix. 3. Support our developers in integrating softphone in iOS and android app. 4. Support our developers in integrating SIP client in hardware embedded devices.
A Python Developer or a team who can develop Desktop Email Client in Python using any Open Source GUI Framework 1. Basic Functionality of Desktop Email Client(Retrieve-Send Mail and Calendar) 2. Support POP3 and IMAP 3. GUI Layout of the Desktop email client should match the give below Sample image.
Hello, I'm looking for a freelancer who knows Asterisk well and can install it on a new VPS for IVR/SIP/VOIP calls after the installation, we'll set the SIP and try to call and recive phone calls * the VPS is in europe * I can install on the VPS one of the most common OS like Centos and ubunto * I rather to install DirectAdmin on that server