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    8,550 mobile phone java sip application tugasan ditemui, harga dalam USD

    Hello everyone, Looking for sip dialer customizable dialer working with provisionning server for platform android/IOS with IM function and contact sharing. Thanks

    $611 (Avg Bid)
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    10 bida

    I am looking for someone to help me develop a website linked to SIP voip server. Asterik knowledge and website integration with our viop server required.

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    26 bida
    IVR and SMS 4 hari left
    DISAHKAN

    we need the sms and IVR developer for our projects the below details are here: SMS- support the send and receive multi SMPP acc...billing for each transaction log generation filter and rules apply DND (Blocklist Feature) connected with MySql IVR- IVR flow CDR sip account call record billing option the criteria based on portal or application

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    5 bida

    Hi, We 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 give

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    13 bida

    We looking for an experienced Cisco Engineer to help us configure a Cisco 2800 series router to switch VoIP (SIP) traffic

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    20 bida
    IVR Product 1 hari left

    ...can be made , for that we require IVR Developer: Experience building voice portal applications and computer telephony integrations. Proficiency in VoiceXML programming, Core Java/ C#.NET, Avaya Aura Orchestration Designer, Avaya Aura Experience Portal, Lumenvox Provides strong knowledge of Call Center Technologies (Nuance, Genesys, Virtual IVR…) including

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    4 bida

    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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    6 bida
    Asterisk Queue Setup 22 jam left
    DISAHKAN

    ...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 in and will be send to queue to be answered by Agents. User 1: Mobile : +91-XXXXXXXX11 User 2: Mobile : +91-XXXXXXXX12 User 3: Mobile : +91-XXXXXXX...

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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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    3 bida

    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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    17 bida

    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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    1 bida

    Upgrading Cisco 7945G IP Phone to SIP Firmware

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    9 bida

    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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    5 bida

    ...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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    9 bida

    ...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 caller id for campaigns

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    1 bida

    ...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 caller id for campaigns

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    10 bida

    ...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 Android mobile versions

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    8 bida

    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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    10 bida

    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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    6 bida
    VOIP product Tamat 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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    14 bida

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

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    13 bida

    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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    10 bida

    ...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 cann...

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    1 bida

    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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    3 bida

    Kamailio sip proxy server need with custom functions

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    3 bida
    JoyGSMDial Tamat 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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    19 bida

    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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    7 bida

    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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    25 bida

    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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    10 bida
    SIP and CHEW Tamat left

    Its an cafe i want a logo for it

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    11 bida

    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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    25 bida

    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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    3 bida

    ...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 of 2 sample IP Phones (so that we can continue to configure

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    2 bida

    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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    3 bida

    Hello. I am ...messages to external numbers without ringing the phone number. The system I am using is FreePBX v14 with trunks from Twilio. 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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    5 bida

    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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    15 bida

    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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    17 bida

    ...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 [log masuk untuk melihat 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 ...

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    Ditampilkan
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    13 bida

    ...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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    3 bida

    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 (store

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    9 bida

    Dear All, We are ne...to monitor 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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    12 bida

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

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    14 bida

    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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    20 bida

    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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    15 bida

    ...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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    ...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 ([log masuk untuk melihat URL])

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    1 bida

    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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