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    2,000 wengo sip trabalhos encontrados, preços em USD

    Strictly bid if you are familiar with softphone/webrtc and web app Works on major web browsers Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat)

    $217 (Avg Bid)
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    4 ofertas

    I have AWS instant hosting PBX, i would like to authenticate SIP trunk using IP Base. Is there any chance you can assist me. thank you ahead.

    $100 (Avg Bid)
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    1 ofertas

    Paint and Sip startup is looking for someone to do the search engine optimization. We are a little business, engaged in DIY, we would like to be involved in the process (eg. keywords set up). Ohh and our website is:

    $100 (Avg Bid)
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    63 ofertas
    Need Twillio expert Encerrado left

    Looking to transitt SIP-services to Twillio, please response with earlier experience and project using Twillio.

    $21 / hr (Avg Bid)
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    5 ofertas

    I will have the system set up and hardware connected. What I need is someone who can do: 2. Codec will be standalone and not registered to an Call Control (CUCM, WEbex) a. Needs to be able to join others’ Webex, Zooom, etc conferences by dialing the SIP URI, etc. b. Needs to be able to do point 2 point ad-hoc conferences with other Codecs. Encrypted video/audio. Possible Side requirement to integrate with Alexa? Alexa currently controls the Samsung hub that controls the lights in the room.

    $633 (Avg Bid)
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    4 ofertas

    Bid if you are familiar with softphone/webrtc and web app Works on major web browsers Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat)

    $96 (Avg Bid)
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    8 ofertas

    Configuracion de una conexion STM1 con el Operador de Comunicaciones para conectarse a la red de telefonia, y configurar un troncal SIP Voip para poder hacer llamadas telefonicas hacia esa conexion, el equipo a configurar es un Cisco ASR1000 series.

    $50 / hr (Avg Bid)
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    Hi Friend, My mini project i need source code to config integrate Opensource PJSIP using WPF c#. Many tutorial on Github but i duno why im not success build Dll from pjsip project and integrate it into c# project. The project result: - Guide to build PJSIP to DLL using in c# ( i try here but not success ) ...pjsip project and integrate it into c# project. The project result: - Guide to build PJSIP to DLL using in c# ( i try here but not success ) - WPF project import PJSIP DLL/ Wrapper C++ function by Swig to call in c# - Can call function support by PJSIP in c# - Can register and make call from WPF Version pjsip is newest on The SIP infor account i will send later

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    # Background We are the Software Company involving in RealTime Communication like WebRTC, SIP, HLS, DASH. I am looking for the great developer who can write very clean code and learn new things very quickly. # Requirement Fullstack engineers who can solve any problem about coding and communication issue in your sides For this job, we are hiring freelancer only, don't need developers in big agency # Question You must answer questions when you apply to this job. If not, I will reject you immediately 1. When creating new React.js Front Side project, what tools and libraries do you use and why(for example, CSS library, Middleware library)? 2. This is my own sample code for react-native project. Can you suggest how I can improve

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    Mac VoIP Softphone Encerrado left

    Looking for a Mac OS Developer, with experience in VoIP and PJSIP Library. Need to build a custom Softphone that can work with our SIP Servers. Here will be the Flow: Application will call an API with username and password to get the SIP Credential. Those credentials will be used to configure Softphone to register on phone system. User should be able to Choose Codecs, Add multiple SIP Accounts. We will hire only people/company who has an experience in VoIP and Sogtphone Application for Mac. Must have worked on similar project and must provide reference. Here you can find windows version.

    $601 (Avg Bid)
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    We are a new VOIP and PBX provider and we are looking for a softphone which has all features thts in BRIA and XLITE and can be used in Windows, Mac, Android and iOS devices. Plz place your bid only if you have developed similar application. To award you the project i would need to see the demo of your application.

    $1296 (Avg Bid)
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    2 ofertas

    Hi i need a Flexisip Server Setup & Configuration complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications. I need good service. when install complete then i need complete installing document

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    Hi i need a Flexisip Server Setup & Configuration server to be set up, complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications. when install complete then i need complete installing document

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    IVR SIP client Encerrado left

    I need you to develop some software for me. I would like this software to be developed for Linux. I need asterisks downloaded and set up on my server. I would like my software to perform following actions; - auto Call - text to speech, with access to change script - dtmf - record voice - send results to telegram

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    We have an asterisk server at our HQ and we have sip phones in our branch offices. We want to Develop a SIP softphone that we customize with our brand and users can install in their windows, mac, ipad, android etc....We need to do voice and video calls with the softphone.....something like LinePhone and Zoiper.

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    11 ofertas

    hello, we are looking for cgrates expert developer who can develop cgrates with our existing open sip server. only experts are welcome on this project. Thanks

    $670 (Avg Bid)
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    5 ofertas

    I have a VPS server that is set up to connect to a few VOIP providers. One of my VOIP providers recently changed the SIP and I need to update my VPS along with the user/pw permissions. My VPS is running: Ubuntu 16.04 (64 bits) I need this done QUICKLY.

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    1) We need developer for softphone client for Windows, Linux and Mac OS X with SIP and IAX capabilities. IAX and SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure. Similar product available is Zoiper (2) We require application for management of PBX customer. We are currently using Issabel. In this application, we require the management under (a) Super User i.e. company and (b) User: Business Partner. Extension and Phone configuration i.e. users creation can be managed by this application as we do in Hardcore PBX software: Issabel. Activation and deactivation, renewals can be managed through this application. Funds allocation for management of user creation and renewal is to be there in application.

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    I need both ios and android app made. a sip client i can connect to any sip server and use username and password (example apps like linphone). I also need DTMF signals to be received. Also need to enable input of sound as microphone / bluetooth / or pre recorded mp3 files - using a browse function and play and pause

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    Hi there, Hope you are doing well. I noticed that you have previously placed a bid to rebrand linphone. I would like to do the same for Windows and Mac OS client with our company brand and linked to our SIP server. Can you please provide an estimate and timeline for such a job please?

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    ...produktu na rynek we współpracy z odpowiednimi działami w organizacji (R&D, Technical Support, Sales); • Współpracujesz z kluczowymi dostawcami oraz klientami firmy; Wymagane kompetencje: • Masz wyższe wykształcenie techniczne (preferowane elektronika, informatyka, telekomunikacja, elektrotechnika, automatyka); • Masz praktyczne doświadczenia z sieciami IP do aplikacji telekomunikacyjnych (m.in. VOiP, SIP, Asterisk, Trixbox, SNMP); • Komunikujesz się w języku angielskim; • Masz silną orientację na realizację celów; • Jesteś kreatywny, dobrze zorganizowany, samodzielny, jesteś nastawiony na współpracę; Nice to have: • Doświadczenie w pracy jako System Engineer / Product Owner; • Doświadczenie w pracy w meto...

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    ...I've tried this on both iphone and android client. Can any asterisk/freepbx professional provide me a quick solution for a fee? Full log: -- SIP/6566811234-00000005 answered PJSIP/90101-00000005 -- Channel SIP/6566811234-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> -- Channel PJSIP/90101-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> [2020-04-05 13:30:53] WARNING[19800]: res_http_websocket.c:508 ws_safe_read: Web socket closed abruptly -- Removed contact 'sip:1se6p552@;transport=ws' from AOR '90101' due to shutdown == Contact 90101/sip:1se6p552@;transport=ws has been deleted [2020-04-05 13:30:54] WARNI...

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    Urgente
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    I need an Android app. I would like it designed and built. I want to implement PJSIP native library and build function for sending sms and calling using voip server

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    I did already install VoIP system with 5 nodes (please, see detail in attached file) - 02 nodes: dSIProuter (Kamailio, RTPengine, Mysql, Nginx (docker)) - 02 nodes FusionPBX (Did config HA with Postgresql BDR & HAproxy) - 01 node install Homer (For monitoring the SIP, RTP & WebRTC) I need a expert to advise, fix some bugs and optimize this entire pilot platform. Include the following: - Advice on system structure. - Config the WEBRTC to SIP client and server (WS on Kamailio, RTP) - Install TURN server - Configure dSIProuter Cluster (HA) and can load balance traffic to FusionPBX nodes

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    This task is straight forward. I need a iOS linphone app that recieve VoIP push and work in background. Only bid if you have done this before. No time waster. I will need to see evidence you have done this or we start with a proof of concept. Open to other iOS mobile sip client that can work with freeswitch and recieve push. I will provide server detail - i use fusionpbx

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    Our main goal to minimize the BW in client side with good quality of voice . 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 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...

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    Link Phone Trunk to Big blue button Big blue button is installed i need you to create a connection to my German sip provider

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    Hello, I am looking for an expert developer with WebRTC XMPP SIP android and ios app developer and integrating with Raspberry Pi. With 90days Recording feature all the resources required for building the project will be provided.

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    ...increase performance (high availability, low latency) and automation (service delivery, testing...) * Design and management of our SIP interconnections with operators across Europe. * Voice quality monitoring and troubleshooting * Development of internal tools to automate / offer web self-service for most common tasks You are : Passionate about technology and expert in VoIP, you wish to join a fast-paced company and a technical team strongly oriented towards open source projects. As an innovative person by nature, you are eager to learn new technologies and to take up new challenges. You master most of the following technologies and methods : * VoIP protocols (SIP, RTP, WebRTC) * Analysis and investigation (logs, IP frames, pcap traces...) * Shell and Python scripting * ...

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    Hi freelancers, I need one voip expert for urgent basis. YOu should have clear knowledge how to set up voip service for call center. You should have clear idea bout DID, SIP TRUNK and other terms. I want to set up 3CX voip services for my company. YOu have to add toll free numbers to 3cx so staffs can pick up calls from their apps. Thanks again.

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    I am using Elastix 2.5 (CentOS 5.11) with 4 NIC's (1 for WAN and 3 For SIP Trunk using IP Authentication) eth0 - WAN IPADDR=75.XX.XX.162 NETMASK= GATEWAY=75.XX.XX.161 eth1 - SIP TRUNK1 IPADDR=100.XX.XX.198 NETMASK= GATEWAY=100.XX.XX.197 eth2 - SIP TRUNK2 IPADDR=100.XX.XX.194 NETMASK= GATEWAY=100.XX.XX.193 eth3 - SIP TRUNK3 IPADDR=100.XX.XX.186 NETMASK= GATEWAY=100.XX.XX.185 all the SIP trunk has to pass the calls 100.XX.XX.5 for authentication & 100.XX.XX.4 for media netstat -rn Kernel IP routing table Destination Gateway Genmask Flags MSS Window irtt Iface 100.XX.XX.5 100.XX.XX.193 UGH 0 0 0 eth2 100.XX.XX.5 100.XX.XX.185 UGH 0 0

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    WebRTC Project Encerrado left

    Works on Chrome, Firefox, IE, Safari, Opera and Bowser Audio / Video call Screen/Desktop sharing from Chrome to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer Multi-line and multi-account Dual-tone multi-frequency signaling (DTMF) using SIP INFO Click-to-Call SIP TelePresence (Video Group chat) 3GPP IMS standards

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    asterisk setup Encerrado left

    ...someone who will build the server from scratch. Not just copy the sample files and fiddle with them. i require the following functionality. 1. Twenty Extensions. SIP Phone and Softphones etc 2. Conference bridge with IVR to Choose a room. Must support Video and Messages. 3. Short text message between extensions. 4. Video Calling between extensions. 5. IVR with Time of day manu for office and none office. Dial by name eyc 6. voicemail for all users. 7. outbound dial for two factor authentication. needs to call end user. ask question and receive response. (details from database ) 8. 4 Analogue Lines. Digium Card installed 9. 2 SIP trunks. Linked to group of extensions. 10. hunt groups. 11. Connection to Microsoft Teams (-optional for discussion) 12. Inbound Outbond ...

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    ...application that runs on Android phones (4.0 or above) and can receive calls through SIP 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 are unable to be met, please let me know before we make a deal. General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Convert audio from/to SIP and GSM networks - Able to run on Andro...

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    Trophy icon Sip-n-Cycle Pedal Cruise Encerrado left

    I am a boat captain in need of a logo for my new pedal boat tour business. Cycle a boat down the river, drinking and having fun with friends. There are several examples like and FUN FUN FUN I would love to have some sort of drinking image ... a toast or pint of beer in the logo.

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    Garantido
    $100
    97 inscrições

    We have an existing RoR/Ember.js application that is in need of on-going maintenance/bugfixes as well as frequent extension for new additions / changes. Split is 85% javascript with ruby only being needed to understand overall architecture and occasionally extend the API for frontend to consume. Exposure to SIP/WebRTC/Websockets is highly desirable as well as there are realtime components of the projects.

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    Video Portal Encerrado left

    We are using SIP/PBX setup (Kamailio-Freeswitch) with Kazoo. We need to create a User page where 2 or more people can have video conference with least settings. Currently Phone and Fax is working. I think video work on certain apps.

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    ...customized dialer for VOIP app to app and app to PSTN calls. Subscribers will sign up online and then use username and password in the app to log in. I am looking for someone to use an SDK library like liblinphone to design the VOIP dialer app. It's OK if you can do it with any other library that is not libliphone as long as the job is done. The app will be for Androis and iOS. I want GPS and SIP headers to be added to the app. The GPS will send location of caller to server because I don't want, for example, a caller in Kenya making app to PSTN calls. Since the app is going to be used in different countries, some countries are restricted from certain operations (like app to PSTN). The GPS is to be used to determine location. The app will also be used to serve coun...

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    We are looking to own an open source(preferably) SIP media server and stream audio on it. This has to receive audio stream from asterisk. This has to start automatically when conference starts up on asterisk (our pbx). We have the PBX'es, just need that audio stream / internet radio that can be accessed by those who can't call in.

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    1: Install and setup GOautodial on server 2: Configure magicjack SIP accounts 3: Test with small group and Update I would like this to be ready for testing on Friday.

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    1: Install and setup GOautodial on server 2: Configure magicjack SIP accounts 3: Test with small group and Update I would like this to be ready for testing on Friday.

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    ...stuff about donuts to bring a smile or two. A joke, poem, limerick, cartoon, video, selfie, etc. The best of the best we will pay $15.00 for. The rest we will give you a byline and show the world how to feel good. We hope this will help a few people forget about the problems of the day and feel good for a few minutes. By the way a nice story about a donut would be good to sit back eat a donut, sip your favorite drink while enjoying a good read....

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    we are a company in need of help configuring and customizing goautodial 4.0 over ...goautodial/vicidial tp see if we are going to use it, if so we will require a lot more help moving forward. below is more details about what we need done I am currently running goautodial v4.0 on AWS https://aws.amazon.com/marketplace/pp/B0859577JD/ref=portal_asin_url I would like help configuring the following - AWS port set up - Dialer (currently when clicking on it is not responding) - Use a 3rd party sip dialer example microsip instead of the build in dialer - Carrier, I will provide you with the information needed - Server ( I need a good explanation on how that is used) - Campaign configuration, Help test the full cycle and explaining some of the features such as survey campaign and bro...

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    Hi We have ASTPP billing installed and configured as a gateway (resource) for our VoIP system, currently, ASTTP uses IP Authentication when we define a client, this result in an issue that all traffic comes from our FreeSWITCH p...when we define a client, this result in an issue that all traffic comes from our FreeSWITCH pointing to this ASTPP (gateway) is considered as 1 client based on that IP and the balance is deducted from one customer no matter the user who is making the call. We want to disabled IP Authentication, or maybe leave it (doesnt matter), but to make ASTPP understand who is the user making the call based on SIP header, and match it with the customers defined in ASTPP system, then check the balance and deduct from it. If you think you can help, please PM me for fu...

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    PHP, MySQL Project Encerrado left

    ...hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. Features needed Connection by phone (SIP trunk) This will be a SIP number that terminates on asterisk (freepbx) IVR which will request conference room number and pin based on the information (numeric only) provided by the script. This will require using asterisks API or .call file. Interface Main interface for users should be controlled by Username and password User should be able to create conferences, schedule by time, and so on. Create a free account

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    Bid if you are familiar with softphone and web apps

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    I have an Android app ( SIP dialer ) I need to add a credit + ads So that the user can earn credit by watching ads and use this credit for us wing the dialer

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    ...looking for some advice technical assist with an issue that we are having with our asterisk based PBX. We have been running an asterisk setup on our internal network for several years with almost no incident. We have two ISDN lines that connect through a B410 digium card to a wazo based asterisk box. Due to us now needing to work remotely we are running into issues. Everyone has their own physical SIP based phone at home that connect to our internal work network. This network has a public IP so there are no issues there. We have also set it up with the following IP rules: 5060:5160/udp ALLOW 10000:65535/udp ALLOW I have got the phones mostly working but I seem to be running into intermittent issues. These issues are as follows:

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    ...Recipt Customize payroll template - Customize Template ready. * For Header and Footer can be upload * cause different Branch to use... and the payroll and report can be gerenate to PDF or can send email to staff. HR Letter - Application form, confirmation letter, extend probation letter, increment letter, promotion letter, termination letter, warning letter. government report - EPF >SOSCO >EIS/SIP >Income-Tax >HRDF management report - Payslip, Payroll report, loan report, claim report, leave report employee report And few more for view the sample can go this site : payroll .autocountsoft .com ** this site for first only use the local host to testing the site.. and i will give the ac host if the site is ready to live. that all. interest please do Message us to...

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