Send audio using sip client asterisk source code jobs

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    16,239 send audio using sip client asterisk source code 仕事が見つかりました。次の価格: USD
    AsteriskGUI 終了 left

    IP-PBXであるAsteriskの設定画面(GUI)をブラウザー上でのDrag&Dropにて設定できるようにしたい。 参考となるのは以下URL [ログインしてURLを表示]

    $54 / hr (Avg Bid)
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    1 入札

    I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM

    $152 (Avg Bid)
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    5 入札
    Trophy icon PAINTnSIP | DESIGN A LOGO 2 日 left

    Design a logo "PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a..."PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a glass of wine to each student, so we call it paint and sip. We are looking for a fun logo.

    $73 (Avg Bid)
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    51 エントリー

    I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM

    $10 - $30
    $10 - $30
    0 入札

    ...the dialer. -Download Recordings - I can download here records from the server. -Recordings - i can listen here the whole conversation and even edit the resultcode of this client (what i mean with edit is example: From sale i can set it to Negative or canceled) Inbound: -Inbound Monitor - I can see here who is calling for the inbound but i can see

    $2966 (Avg Bid)
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    45 入札
    Asterisk PBX with WebRTC - GURU 6 日 left
    認証完了

    To duplicate an existing Asterisk Server with WebRTC ( HTTP5 - SSL - etc.) onto another machine and make some changes. Works together with an Apache Web Server and a MySQL Data Base, on separate machines. Must function as an immediate - identical backup !

    $966 (Avg Bid)
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    13 入札

    zoho's phonebridge plugin supports up to asterisk 1.4 with asterisk 1.8 partially working (only incoming calls) with modern asterisk versions (11 and up) not working at all do to (apparently ) java problem zoho's support says that those values : dialed channel, linked channel, dialing channel are NULL instead of being filled with information. we

    $227 (Avg Bid)
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    3 入札
    Twilio SIP set-up 5 日 left
    認証完了

    Please set-up Twilio phone number with SIP so I can use on following site: [ログインしてURLを表示]

    $19 (Avg Bid)
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    4 入札

    We are a Philippines Company looking for an expert in Docker, Kubernetes and Google Cloud to help us speed up migrating 3 existing VMs (Java App, Asterisk and Postgres) to Docker Containers deployed on Kubernetes on Google Cloud. We have already containerised the Apps. We now want to accelerate deployment and to then have assistance with "peeling out"

    $21 / hr (Avg Bid)
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    5 入札

    I need a complex asterisk job ... need experts only.

    $15 / hr (Avg Bid)
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    15 入札

    I'm looking for a tech who already...completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid

    $555 (Avg Bid)
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    4 入札
    VOIP - CALL DROPS 4 日 left
    認証完了

    We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.

    $99 (Avg Bid)
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    18 入札
    OBD CALLING 3 日 left

    Hello We have SIP,PRI and want to build script where we can make OBD Calling (asterisk),

    $1647 (Avg Bid)
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    7 入札

    I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system

    $77 (Avg Bid)
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    2 入札

    ...[s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [ログインしてURLを表示]: Caller ID name is '+918820094576' number is '8902050098' This should be a very sma...

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    9 入札

    ...checkout page the fields, Pais, Endreço. Cidade,Estado, CEP, they are [ログインしてURLを表示] I do not know why the red asterisk does not appear. I installed the plugin WooCommerce Checkout Field Editor, although I enter the fields as mandatory, the red asterisk does not appear , and it is not possible cancel the writing (OPTIONAL). mysite: [ログインしてURLを表示]

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    51 入札
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    5 入札

    Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for

    $39 / hr (Avg Bid)
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    10 入札

    ...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed

    $37 / hr (Avg Bid)
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    1 入札

    Looking for someone who can configure Dinstar GSM Gateway to work with my local asterisk setup.

    $28 (Avg Bid)
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    10 入札

    Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.

    $250 (Avg Bid)
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    1 入札

    Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call

    $660 (Avg Bid)
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    26 入札

    ...hours). We offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers;

    $18 / hr (Avg Bid)
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    27 入札

    ...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a

    $120 (Avg Bid)
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    13 入札

    We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization

    $1356 (Avg Bid)
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    29 入札

    App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.

    $631 (Avg Bid)
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    13 入札

    ...require someone hands on experience in installing goautodial or vicidial solution on Google cloud computing. Good understanding of Linux, Asterisk and Vicidial is essential. We wish to start testing Asterisk and Vicidial in the cloud from Google. We require setup, testing and ongoing support. Phase one is setup so we can start testing. Phase 2 will

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    14 入札

    ...outsource for those tasks when needed. Currently i can be specific on a project which you can tell me you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry

    $20 / hr (Avg Bid)
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    1 入札

    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[ログインしてURLを表示]" para provisionamiento remoto.

    $130 (Avg Bid)
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    4 入札

    I am having problems getting a dhadi/Asterisk/POTS configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work properly. Config files and CLI output: [ログインしてURLを表示] Dialplan: [ログインしてURLを表示] The machine once finished will backup and replace my old Asterisk servers that basically operate as POTS

    $175 (Avg Bid)
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    5 入札

    Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.

    $21 / hr (Avg Bid)
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    7 入札
    $191 \u5e73\u5747\u5165\u672d\u984d
    6 入札

    I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer • Blind Transfer • Supervised Transfer...

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    23 入札

    ...if VoIP package has enable, if enabled then execute real-time billing logic with client package rate and remove credit from credit balance. If client’s package has free minutes, check if over minutes then execute real-time process. Issue remarks in Manage Credit/Client Description field within table as follows: (you may change your variable names to

    $609 (Avg Bid)
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    9 入札

    Hey guys, I'm looking for a server...be setup with a hosting server, mail server, SSL for domains, and I will set up an Asterisk server in one VPS which also needs to be secured. I also need help migrating over from my current VPS to the new dedicated server setup. If you think you can do this please send me a message with the word "iluitech" in it.

    $431 (Avg Bid)
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    41 入札

    I need you to develop some software for me. I would like this software to be developed for Linux using PHP. Asterisk voip and e1 integration for call center operator.

    $1254 (Avg Bid)
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    29 入札

    ...as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with a proposal and make sure to write "iluitech" in your message so I know you actually read the requirements and know that you can do it. If you have SIP experience and know PHPAGI it would be a huge plus!

    $592 (Avg Bid)
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    50 入札

    Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[ログインしてURLを表示]' SIP_PORT = 5060 LOCAL_IP = '[ログインしてURLを表示]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [ログインしてURLを表示] (200 OK or 301 Moved Permanently OR 401 Unauthorized etc...) +save

    $143 (Avg Bid)
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    4 入札

    Hello, i want script to Test sip accounts with Back SIP response codes [ログインしてURLを表示] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent

    $115 (Avg Bid)
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    7 入札

    hello, i have this package : [ログインしてURLを表示] need to...package : [ログインしてURLを表示] need to install on my server windows then build api requests to manage users and add sip accounts SO i will be able later to use on my custom cms

    $90 (Avg Bid)
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    5 入札

    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

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    10 入札

    VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist

    $158 (Avg Bid)
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    11 入札

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [ログインしてURLを表示] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [ログインしてURLを表示] will be long term project if satisfied with the [ログインしてURLを表示] budget is 100$ for this project

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    13 入札

    Voglio un server voip che giri con asterisk

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    1 入札

    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [ログインしてURLを表示] /hangupall [ログインしてURLを表示] /answer microsi...[ログインしてURLを表示] msip:micdown - Speaker volume up: [ログインしてURLを表示] msip:speakup - Speaker volume down: [ログインしてURLを表示] msip:speakdown Change header logo You must provide the new source code and point the changes made

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    4 入札

    hello, please check this doc [ログインしてURLを表示] i need code to test my back Udp + Tcp connection to my sip server with show error Error = { TRYING = 100, RING = 180, TIMEOUT = 408, BUSY = 486, DECLINE = 603, OK = 200, UNAUTHORIZED = 401, FORBIDDEN = 403, NOTFOUND = 404, PROXY_AUTH_REQUIRED =

    $38 (Avg Bid)
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    2 入札

    Need Linux guy for Ubuntu for fixing some issues with Asterisk server

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    22 入札
    VOIP/SIP Project 終了 left

    ~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at...(Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).

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    9 入札

    Hi kristenhutchiso8, I noticed your profile and would like to offer you my project. We would like to hire you to set up a new catalog upload to SIP with existing iProd iPrice and iPhoto database files to include @ 160 products on GSA Advantage with our new GSA contract. This is the first of 4 projects with GSA Advantage and FEDMALL we would like to

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    1 入札