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    8,543 java sip client iphone công việc được tìm thấy, giá USD

    Mình cần hỗ trợ config Elastix, inbound, outbound và sip trunk

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    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.

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    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

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    Upgrading Cisco 7945G IP Phone to SIP Firmware

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    I am providing vas on IVR by using java platform, so I need a java coder who as I need to get some changes done in coding , add new product and also create a SIP for OBD

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    Cold-caller for motion design studio 1 ngày left
    ĐÃ XÁC THỰC

    ...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:

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    ...outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider caller id for campaigns and run DID NUMBERS UPLOAD FOR INCOMING CALLS ROUTE THEN WITH IVR /CALL CENTER IP SIP/ REROUTE OPTION Also Can create SIP account for receive incoming calls agents so when did routed to sip agent calls

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    ...outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider caller id for campaigns and run DID NUMBERS UPLOAD FOR INCOMING CALLS ROUTE THEN WITH IVR /CALL CENTER IP SIP/ REROUTE OPTION Also Can create SIP account for receive incoming calls agents so when did routed to sip agent calls

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    SIP to Whatsapp gateway 2 giờ left
    ĐÃ XÁC THỰC

    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

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    Hi Tran H., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I saw you were awarded to this project "SIP to Whatsapp gateway" did you finish it?

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    Build Google Data Studio dashboard Sắp kết thúc left
    ĐÃ XÁC THỰC

    Want to build a google data studio dashboard displaying phone extension statuses, sip provider statuses, and outages for digium phone system.

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    Setup OpenSIPS VoIP server for Termination Đã kết thúc left
    ĐÃ XÁC THỰC

    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

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    VOIP product Đã kết thúc left

    ...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

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    Integrate Issabel-Asterisk with Twillio Elastic Sip Trunking to records Calls from Twillio (through Flex)

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    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.

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    Freepbx one-way audio trouble Đã kết thúc left

    ...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

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    I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.

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    SIP Proxy Server Đã kết thúc left

    Kamailio sip proxy server need with custom functions

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    JoyGSMDial Đã kết thúc left

    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

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    Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .

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    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    Dear all, We have a urgent Vicidial system to se...information and agent disposition data into one outlook. That means initial upload of list information must converge with output of agent disposition to create one report 6) To create sip trunk for testing 7) To create sample lead list 8) Testing and commissioning 9) Agent and supervisor training

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    SIP and CHEW Đã kết thúc left

    Its an cafe i want a logo for it

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    SIP dialer for Android and IOS Đã kết thúc left

    SIP dialer for Android and IOS. Should support most Android and IOS versions including IOS version 12. Features: Runs behind NAT Support Bluetooth Echo Cancellation Push notification Call history display Account balance display Address book integration Support UDP and TCP protocols Support users to Login or Signup Supported languages: English, Arabic

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    Need an Asterisk/FreePBX guru to show me how to setup xlite and Microsip sip phones to register with FreePBX extension.

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    ...basic IVR. Only one level. 2. Setup of 2 profiles of users. a. Will be able to make local calls only b. Will be able to make local and external calls 3. Configuration of 2 SIP and DID services 4. Configuration of Conference rooms (phone bridges) 5. Configuration of VOIP VLANat the IPPBX and Phone level (network switch is already configured) 6. Configuration

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    Create IVR on Raspberry PI Đã kết thúc left

    i need someone to create and IVR for me using Asterisk that will interact with mysql database for callers details and readout details form database using TTS (Hindi ...done it using PHP api but never got time to finish this (i don't have the codes now) there are 3 levels of menu and it must use 3G datacards to receive make calls not just sip or fxo

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    Voice Mail Broadcasting Đã kết thúc left

    Hello. I am looking for someone that can build a system that will deliver voicemail messages to external numbers without ringing the phone...This is also knowen as VoiceCasting. It's important that you understand that the phone numbers are NOT phone numbers or extensions within the PBX. Must have vast knowledge of SIP and PBX with FreePBX platform..

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    I want voice engineer to do IP Telephony Cisco call manager installation and expressway server ,so that sip phone register over internet.

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    Rephrase an academic article Đã kết thúc left

    Hello, I have an article with about 50 pages. It talks about VOIP and TOR networks. I need to rephrase this article so it can not be recognized or even a...plagiarism checkers. All the article is in English and rephrasing should be in English too, I guess a technical background is needed. Specially talking about protocols Like RTP, SIP,UDP and VoIP.

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    TEXT VIA WEB APP Đã kết thúc left

    ...The client may receive some visual and auditory signaling that a new text message or photo has arrived We need a control panel to create the client, suspend the client, with its respective fields such as client name, authorized telephone number, text records, API field and other required for the functionality of the WEB application. The client can

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    Nồi Bật
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    configure Kamailio with ipv6 Đã kết thúc left

    ...and register to IPv6 sip servers 2.) IPv6 clients should be able to use the Kamailio server as a proxy and register to IPv4 sip servers 3.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv4 sip servers 4.) IPv4 clients should be able to use the Kamailio server as a proxy and register to IPv6 sip servers 5.) TLS must

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    SIP Client with Yii2 Đã kết thúc left

    ...framework in cloud (with public IP) and I want to integrate it with digital PABX and providing SIP client under Yii2 framework. The PABX has set up an SIP server which only has local/private IP address and I want to add SIP client within my web apps. The SIP client should recognize incoming call numbers (store the incoming number, extension and t...

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    Dear All, We are need a servic...4 asterisk machine and provide event logs and alert on the following status of the machine, IP phones, line and bandwidth below. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Please submit your proposal by 25th of January 2019. Thank you, John

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    Hi we are looking for the code of a webrtc button that can make a sip call(audio only)( [đăng nhập để xem URL] maybe) be compatible with the mayor browsers at moment([đăng nhập để xem URL]) and we should config stun and turn username and password sip server(asterisk) and number that will be called

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    voip to Native Phone dialer app Đã kết thúc left

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for

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    Android SIP Client Đã kết thúc left

    SIP Client Application for Both IOS and Android prefer to go with Linphone Features : - Audio & HD video calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Instant Messaging with message delivery status (IMDN) Pictures and files sharing Contact list Call History Display of advanced call

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    Website with good UX and UI Đã kết thúc left

    ...BSE (Bombay stock exchange) API for buying and selling MF in safe and secure manner using the BSE payment gateway (through third party vendor) which can be assessed through client login ID password (through a sub domain). We just want good UX & UI website with fast loading website and all features of SEO and GOOGLE friendly features so that it can boost

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    Tools and Calculators dev Đã kết thúc left

    This project on creating Tools and Calculators for an Indian financial website. Examples of tools will be of Home Loan calculator, SIP calculator etc Please share relevant work references

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    ...written in Java. It includes support to monitor SIP devices registered to and asterisk server and it does it by connecting to Asterisk's AMI interface and querying for SIP registrations. If the list of registrations change, OpenNMS generates an event or an alarm. OpenNMS depends on a old version of asterisk-java ([đăng nhập để xem URL])

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    Asterisk and Queuementrics Đã kết thúc left

    Good day, I am looking for a resource to configure queue metrics and freepbx, this is a new clean setup, new boxes. This you will start with clean setup. We have already a sip trunk provider and there accounts need to be setup also Both boxes is already loaded, we only nee to finish off the setup and configuration. We wish to extent both products

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    Free PBX inbound route Đã kết thúc left

    If possible, I would like to route an inbound route based on TRUNK and NOT based on DID. FYI: My SIP provoider is NOT sending the DID, so the routing should go based on TRUNK (thus when an incomming call on trunk A comes, then it should be routed to extension 200, when an incomming call is comming on trunk B, it should be routed to extension 201 ect

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    Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.

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    Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

    $282 (Avg Bid)
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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

    $132 (Avg Bid)
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    2 lượt đặt giá

    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    AXNET VoIP Đã kết thúc left

    I am Peter from Rwanda/ Africa. I am looking someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones

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    Hello, We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough

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    I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.

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