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    16,414 mobile softphone symbian sip voip pekerjaan ditemukan, seharga USD

    Memindahkan database dan script yang telah terinstall di server centos 7 yaitu menggunakan asterisk. Software telah tercustom akan dipindahkan ke server baru. Diharapakan yang telah memahami asterisk dan freepbx

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    Softphone android Berakhir left

    Un softphone que maneje cuentas SIP para ANDROID

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    Saya Pernah Direkrut Ngentry Dikantor Badan Stastistik Itu Terjadi Tahun 2015 . Jam Ngentrynya Dimulai Jam 12 Siang Sampai Jam 4 Pagi , Kebetulan Saya Sip Jam 9 Malam Sampai Jam 4 Pagi .

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    ...Berikut keterangan fitur yang perlu di modifikasi. Modifikasi Aplikasi Anroid ( VoIP ) 1. Memodifikasi agar Aplikasi VoIP tetap berjalan di Running Background, Walaupun aplikasi tertutup. 2. Menyaring nomor kontak yang akan ditampilkan hanya yang terdaftar di server ( VoIP ). 3. Memberikan database pada aplikasi android agar bisa menyimpan nomor yang

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    Membuat voip android application memgunakan android studio untuk membuat [login untuk melihat URL] database serta mengunakam asterisl sebagai server

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    setting IP PBX Berakhir left

    jasa setting untuk keperluan kami yaitu: Billing Syste...System / Budgeting per ID / User Recording All Call IVR / Auto Attendance Call Fax to Mail & Mail to Fax Bisa Cabang IP PBX Yealink BLF [Busy Lamp Field] Sip Messenger connect Android APPS VOIP saat ini kami memakai program elastix dengan hardware openvox a814hj0582xk dengan chip 8fxo

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    ...Care untuk bekerjasama dengan PT Kronus Indonesia - Perawat, Fisioterapi, Okupasi Terapi, Terapi Wicara & Bidan - Laki-laki dan Perempuan - Usia 21th – 45th - Memiliki STR/SIP yang masih berlaku - Mampu bekerjasama dengan Team / Independent - Penempatan wilayah Jakarta dan Luar Jakarta (Bandung, Semarang, Yogyakarta, Surabaya, Medan, Makassar atau

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    this program for information tcooperation in Goverment "Pemerentah kabupaten WONOGIRI"(Dinas Koperasi dan UMKM) INDONESIA, Can use for ALL Dinas koperasi and UMKM ind Indonesia

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    HARJONO file Berakhir left

    ...Blackberry, Nokia, dan merek lainnya :) , Apa itu Android Android adalah sistem operasi yang digunakan di smartphone dan juga tablet PC. Fungsinya sama seperti sistem operasi Symbian di Nokia, iOS di Apple dan BlackBerry OS. Android tidak terikat ke satu merek Handphone saja, beberapa vendor terkenal yang sudah memakai Android antara lain Samsung , Sony

    $182 - $548
    $182 - $548
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    IVR and SMS 6 Hari left
    TERVERIFIKASI

    we need the sms and IVR developer for our projects the below details are here: SMS- suppor...send and receive multi SMPP account 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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    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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    I want the page to be on Top of Google without Adverts. Linkbuilding, SEO Word setting, Forum postings, traffic leading, Facebook page etc.. Pls write suggestions how you going to make it no. one?

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    VOIP SWITCH 4 Hari left
    TERVERIFIKASI

    Need help configured my new voipswitch platform

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    WE WOULD LIKE SOME OF OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME ,...OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME , CISCO SPA 504G PHONES. THEY NEED TO BE PROGRAMMED. ALL TOGETHER WE HAVE 25 PHONES THAT NEEDS TO PROGRAMMED FOR VOIP. YOU MUST BE FAMILIAR WITH MITEL PHONES, IF NOT THEN PLEASE DON'T BID FOR THE WORK

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    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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    IVR Product 3 Hari left

    ...Maintain and manage Genesys Routing, Framework and reporting. Responsible for supporting call 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

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    need a softphone coded to dial out automatically

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    need to set up a softphone that can be on a continuous loop to dial out on incoming data(telephone #s). I have incoming data that includes a phone # within the data. I have it set with my twilo account to call out on the phone # instant but i need a softphone installed so it can call out on the phone # automatically.

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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 2 Hari left
    TERVERIFIKASI

    ...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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    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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    Find me some Leads for a VoIP Service. The Company offers cheap rates for VoIP calls via Mobile App (Andorid, Symbian, IPhone). Please, I do not need Email Addresses to send bulk Emails. I need some traffic twoards my site. And acctual leads that generating paying customer. Any suggestions appreciated.

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    Tenho um projeto de plataforma Saas que deve conter um discador/softphone/webphone que preferencialmente deve ser desenvolvido em JS e que se comunique ou seja faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. A...

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    Asterisck pbx server installation (fusion or frepbx), configure and integrate with Odoo VoIP module v10 for Web based calling using WebRTC protocol. Need proven or similar project experience in Odoo platform.

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    Mantenimiento y nuevos proyectos

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    voip control Berakhir left

    voip control management and vpn

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    Develop a whatsapp equivalent mobile app using webrtc for voip and text + required backend.

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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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    VOIP Calling Berakhir left

    I need a software that allows me to make phone calls on the internet from a free VOIP service. You must find the free VOIP service for me. I should be able to change the phone number from which the calls are made to random numbers each time a phone call is made. I prefer the code in VB.net but if you can not do it in VB.net please explain which coding

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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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    Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload

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    ...only bid with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details

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    ...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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    Project for Tran H. Berakhir left

    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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    laravel expert Berakhir left

    ...Customer accounts Create extension and assign in Queue in FreePBX Add, edit, delete Customer messages View call logs Softphone pre provision (softphone setup - not dev) Agent - Features Answer calls (with softphone) - done with softphone Be presented with a screenpop for incoming call - company info + forms (messages) if assigned Place notes on specific

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    VOIP product Berakhir 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 a stand alone person who runs a FreePbx and A2billing platform I need android / IOS / and pc dialer that works via my platform.

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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 Berakhir left

    Kamailio sip proxy server need with custom functions

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    JoyGSMDial Berakhir 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 Csipsimple to include the following features. - 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

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