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    8,462 free sip flash media server робіт знайдено, ціни вказані в USD

    Що потрібно… «Розвернути» на сервері ASTERISK Налаштувати транк sip gsm +transfer Запис розмов та фіксацію номерів трансферних Адмін панель Панель доступу user Надаштування GSM шлюзів

    $576 (Avg Bid)
    $576 Сер. заявка
    5 заявки
    SIP register with Custom software 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [увійдіть, щоб побачити URL] i want to run this program on a router Router Link [увійдіть, щоб побачити URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580 Flash MB:

    $555 (Avg Bid)
    $555 Сер. заявка
    1 заявки
    Programming Bandwith.com 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    Help programming sip endpoints, applications voice & sms in [увійдіть, щоб побачити URL] I need to connect my [увійдіть, щоб побачити URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [увійдіть, щоб побачити URL] Prior experience

    $5 / hr (Avg Bid)
    $5 / hr Сер. заявка
    2 заявки
    Multichannel Phone Dialer 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

    $544 (Avg Bid)
    $544 Сер. заявка
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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

    $152 (Avg Bid)
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    sip to Native dialer APP 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    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 this

    $510 (Avg Bid)
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    SIP 2 SIP call optimization bandwidth 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i s...

    $838 (Avg Bid)
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    VOIP, SIP Expert 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

    $206 (Avg Bid)
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    Connect IP PBX to Twilio WebRTC 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

    $42 (Avg Bid)
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    5 заявки
    SIP Proxy Server install 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

    $193 (Avg Bid)
    $193 Сер. заявка
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    skype for business on-premise installing and setting up 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

    $250 (Avg Bid)
    $250 Сер. заявка
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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Secure Issabel PBX (asterisk) 2 дні(-в) left
    ПІДТВЕРДЖЕНО

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

    $65 (Avg Bid)
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    Trophy icon Création logo Startup 2 дні(-в) left

    ...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

    $56 (Avg Bid)
    Гарантований
    $56
    85 робіт

    ...and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both parties upon pick up 3) To store voice messages on my server (in wav or Mp3), email notification when message is received and ability to ftp download message Some of the existing systems: [увійдіть, щоб побачити URL] , [увійдіть, щоб побачити URL]

    $335 (Avg Bid)
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    Turn a led on, when a button is pressed on a Yealink SIP-T42S 21 годин(-и) left
    ПІДТВЕРДЖЕНО

    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

    $324 (Avg Bid)
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    require a AVAYA Exchange PHP Scripts 4 годин(-и) left
    ПІДТВЕРДЖЕНО

    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

    $181 (Avg Bid)
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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

    $25 (Avg Bid)
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    3 заявки

    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

    $158 (Avg Bid)
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    VOIP, SIP, cyber security, Networks. Закінчився left

    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

    $52 / hr (Avg Bid)
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    sipxcom sip system setup Закінчився left

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

    $26 (Avg Bid)
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    I need logo and letterhead designer Закінчився left

    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

    $281 (Avg Bid)
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    VOIP ASSISTANCE Закінчився left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Trophy icon JT&T Brand 4 дні(-в) left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

    $220 (Avg Bid)
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    $220
    516 робіт

    ...the below requirement MY REQUIREMENTS 1) I have to shift my existing asterisk vicidial server files ( my dialplans, modules , database ,IVR , [увійдіть, щоб побачити URL] , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to new & better hardware /server Xeon processor 3.2 GHz with latest Asterisk & Vicidial software installed 2) Sugar CRM Com...

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    Install freepbx on CentOS machine Закінчився left

    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    Project for Prostitov A. Закінчився left

    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [увійдіть, щоб побачити URL] secret GZD397XU Host [увійдіть, щоб побачити URL] port 5060 Register String 390415541985@[увійдіть, щоб побачити URL]:GZD397XU:390415541985@[увійдіть, щоб побачити URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [увійдіт...

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    IAX SIP VOIP Softphone Закінчився left

    This project is only for those freelancer that has existing similar app to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications

    $2570 (Avg Bid)
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    HSMWorks-sip panels/ Panels Закінчився left

    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

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    QT based SIP Client with Dynamic CRM Закінчився left

    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    VOIP, SIP, Cyber Security Закінчився left

    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

    $14 / hr (Avg Bid)
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    Avaya VOIP Phone System Help Закінчився left

    I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...

    $242 (Avg Bid)
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    install sip on cisco 7960 phones Закінчився left

    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

    $207 (Avg Bid)
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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

    $139 (Avg Bid)
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    Configure FreePBX/Asterisk with NAT Закінчився left

    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

    $109 (Avg Bid)
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    Voicemail Drop Windows Software -- 2 Закінчився left

    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [увійдіть, щоб побачити URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

    $202 (Avg Bid)
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    Predictive Dialer Закінчився left

    we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list

    $458 (Avg Bid)
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    SIP to GSM Gateway for Android - New Закінчився left

    Hi I need an application that can receive a call 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. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default

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    ...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-

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    Implement SMS verification in VoIP softphone and php backend, create IOS and Android softphone General project description: We are running an A2Billing + Asterisk server and want to have a branded Softphone for Android and IOS. Registration and provisioning of softphones must happen with SMS verification. For the softphone we want to use the source

    $642 (Avg Bid)
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    multi asterisk in one server Закінчився left

    ...asterisk server multi useing like i have 10IP they copy asterisk folder 10 different name and run asterisk 10time its works file 10 asterisk server so i want to use same way i have 5ip i want to run 5asterisk server with different name and connect with different database 1main server other all asterisk server work like VM host1 IP [увійдіть, щоб побачити URL]...

    $225 (Avg Bid)
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    Bitrix24 telephony integration -- 2 Закінчився left

    Hello All, We need to integrate Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking

    $18 (Avg Bid)
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    We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Proficient in Lantek program (lazer tube cutting) is a must. We have ongoing work for the right programmer on a long term basis.

    $250 - $750
    $250 - $750
    0 заявки

    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Artificial Intelligence: It involves Studying Human thought processes representing thought process on machines. Knowledge encompasses the implicit and explicit restriction

    $3 / hr (Avg Bid)
    $3 / hr Сер. заявка
    2 заявки

    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

    $976 (Avg Bid)
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    7 заявки
    URL Free to Call System Закінчився left

    URL Free to Call System from the website, to use my sip trunk for clients to call me free.

    $43 (Avg Bid)
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    ...When number clicked on "Free Call" button, user get connecting message and generate DTMF button on screen so user can press options from IVR and user use inbuilt microphone then connect call to SIP address on Incredible PBX OR When number clicked on "Free Call" button, viewer types in his/her phone number and submits to SIP address which setup...

    $204 (Avg Bid)
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    ccnp voice Закінчився left

    ccnp voice training and configuration for voice gateway mgcl sip and h323

    $109 (Avg Bid)
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    2 заявки
    Softphone in Android and IOS Закінчився left

    I need an app for Android and IOS that basically takes and makes sip calls. It can be an open source phone labeled with the name that I need.

    $191 (Avg Bid)
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    12 заявки