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    8,545 create web based sip application iş bulundu, ücretlendirmeleri EUR

    Hi thepasserby82 Merhaba Ankara'da bir şirketiz.,Bir VoIP-SIP programı geliştirmek istiyoruz. Farklı olarak speech to text ile entegre olacak. Bunun örnek kodları github ta mevcut. Bunu imkan veren en iyi voip open source çözümü olarak asteriks bulduk. Türkiye de biri ile çalışmak bizim için daha avantalı olacaktır. İletişim bilgilerini pay...

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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

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    Belirli bir prefixle ayrıştırılmış belirli bir IP den H323 ve/veya SIP protokollerinden gelen toplu çağrıları elemek için whitelist ve blacklist yazılımının oluşturulması. Whitelist ve blacklist te bulk ve/veya tek tek data girişi için kullanıcı ara yüzü oluşturulması. Filitreden geçen trafiği tekrar ilgili prefix ile belirli bir IP ye yönlendirilm...

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    IVR Product 6 gün left

    ...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 and maintain Routing Applications and Configuration Objects. Understands the Genesys

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    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

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    Asterisk Queue Setup 6 gün left
    ONAYLI

    ...+91-XXXXXXXXXX(Agent or user) come to my SIP and thus to my dialplan. If Agent calls he should be loged in as an agent If user calls he should be transferred to a Logged in user. Rewording the above requirement. I have my Agents say A1,A2,A3 on remote location. I have my callers C1,C2,C3,C4,C5 too on remote location. I have a SIP installed on my server which

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    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

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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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    ...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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    ...chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload numbers direct form text file sime only numbers and also can generate numbers and save them into phone book add provider for outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider

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    ...chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload numbers direct form text file sime only numbers and also can generate numbers and save them into phone book add provider for outbound calls IP/SIP add provider for inbound calls IP/SIP Dialer where can select phone book survey wait time provider

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    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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    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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    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 Bitti 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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    ...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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    Kamailio sip proxy server need with custom functions

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    JoyGSMDial Bitti 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 Bitti left

    Its an cafe i want a logo for it

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    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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    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 and English) there should be option to call http apis and also the call flow should be easy to modify i have previously done it using PHP api but never got time to finish this (i don't

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    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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    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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    ...America. I am in need of developing a web application so that my clients can do the following: To be able to send and receive texts using the platform of [login to view URL] I seek to make an application as if it were a real mobile phone that can send and receive the texts and photos in the same WEB application for any platform. You can use this as a

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    Özellikli
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    ...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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    I have a web application built with Yii2 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 (stor...

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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)( [login to view URL] maybe) be compatible with the mayor browsers at moment([login to view URL]) and we should config stun and turn username and password sip server(asterisk) and number that will be called

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    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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    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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    ...The features/ pages would include 15-30 Pages around. NAV of MF linking for fetching Price (NAV) info (we already have the membership of BSE). Retirement Planning Calculator SIP Calculator other calculators with various drop down Home page: Scroller images Corner: Login call us icon for: request a call back with preferred time slot & day 1. About us:

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    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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    ...com/OpenNMS/opennms) is an open source monitoring and systems management platform 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

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    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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    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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