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    8,223 sip phone client windows mobile 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)
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    5 заявки
    VOIP - CALL DROPS 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    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.

    $105 (Avg Bid)
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    13 заявки
    OBD CALLING 5 дні(-в) left

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

    $1663 (Avg Bid)
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    6 заявки
    Install Vicidial to an existing FreePBX system. 5 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...VoIP provider. I gave the green color for your easier to understand. -- Executing [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 [увійдіть, щоб побачити URL]: Caller I...

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    1 заявки
    OpenVOX GSM gateway configuration changes. 4 дні(-в) left
    ПІДТВЕРДЖЕНО

    ...provider. I gave the green color for your easier to understand. -- Executing [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 [увійдіть, щоб побачити URL]: Caller I...

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    7 заявки
    Project for Bhavin B. 6 дні(-в) left

    ...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 заявки

    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 заявки

    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

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    29 заявки

    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.

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    Provisioning cloud Cisco 7911 Закінчився left

    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[увійдіть, щоб побачити URL]" para provisionamiento remoto.

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

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    Delphi sip client Закінчився left

    I want to get Delphi sip client. delphi version is 7

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    6 заявки

    ...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

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    9 заявки

    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own ...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!

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    Python sip registering script Закінчився left

    Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[увійдіть, щоб побачити URL]' SIP_PORT = 5060 LOCAL_IP = '[увійдіть, щоб побачити URL]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [увійдіть, щоб побачити URL] (200 OK or 301 Moved Permanently OR 401 Unauthoriz...

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    Voip sip registering script Закінчився left

    Hello, i want script to Test sip accounts with Back SIP response codes [увійдіть, щоб побачити URL] 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

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    7 заявки

    hello, i have this package : [увійдіть, щоб побачити URL] 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

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    5 заявки

    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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    VLC and FreePBX on Ubuntu Закінчився left

    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

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    Looking For Android sip app Devloper Закінчився left

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [увійдіть, щоб побачити URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [увійдіть, щоб побачити URL] will be long term project if satisfied with the [увійдіть, щоб побачити URL] b...

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    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [увійдіть, щоб побачити URL] /hangupall [увійдіть, щоб побачити URL] /answer [увійдіть, щоб побачити URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [увійдіть, щоб побачити URL] msip:hangupall [увійдіть, щоб побачити URL] msip:answer [увійдіть, щоб п...

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    want build code using nmap Закінчився left

    hello, please check this doc [увійдіть, щоб побачити URL] 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 =

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    VOIP/SIP Project Закінчився 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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    Project for Kristen H. Закінчився left

    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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    Skype for business Direct Routing Закінчився left

    We are looking for a skype for business office365 consultant to assist in setting up a direct routing environment from the office365 environment to a SIP SBC.

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    Hi There, We require help to build a proof of concept.s We want a system connect a Client to a Carrier that is scalable. However, we do not want to disclose any of the Carrier IPs to the Client and Vice Versa. I've done some research and using OpenSIPS and RTPproxy can help with this but I am having trouble setting this up on AWS. For the proof of

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    I am looki...connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.

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    I'm searching for a multithreaded perl script that calls a phonennumber mutiple times with predefined sip-accounts and 4 additional functions.

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    Sip dialer voice recorder coding Закінчився left

    Sip dialer voice recorder coding Must know about PHP and .net programming. Experienced person.

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    Help setting up Freepbx Закінчився left

    i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me

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    SIP to Local call App Закінчився left

    Need to create a SIP app that can install on Android. Register to SIP server Receive call from SIP server and initiate call via telephone sim card Connect audio from telephone call to SIP call

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines

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    ...However, the client IP phones are behind NAT and common NAT traversal techniques such as STUN and TURN are cannot be used. I am configuring an outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: 1. Perform NAT traversal on any incoming or outgoing SIP connections

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    install FreePBX v.13.0.121+; Закінчився left

    1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

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    Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string

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    SIP Media Gateway Engine for Chatbot Закінчився left

    ...will bring voice to text-based chatbots. The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media

    $26 / hr (Avg Bid)
    Обраний Терміновий Угода про нерозголошення
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    linphone + uvc camera Закінчився left

    I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.

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    ...app) To be up to date and to attract more users, we are going to assign a dedicated phone number to every registered user so they can call orher app members or other non app numbers, they would also be able to receive call from any phone or device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP)

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    Asterisk Black listing Закінчився left

    We need a solution in asterisk that help us block incoming calls with the same numbers (robot calls). We are star...if any new call is coming in, then look into the Database to see if the incoming number is in the database, and if it is, then just block the call and sent a "486 Busy Here" Sip response, otherwise asterisk can place the call as usual.

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    Android with SIP/VoIP caller program Закінчився left

    ...develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module More details (ask) ////////

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    ...APPBuilder that allows to include the SIP / VOIP SDK to connect to ASTERISK as an option and if it were not possible in ANDROID Studio The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module

    $30 - $250
    $30 - $250
    0 заявки
    Trophy icon Content Writing Закінчився left

    ... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class

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    Гарантований
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    Paint and sip for kids Закінчився left

    looking for an artist/painter/instructor to teach kids and art classes/parties for different ages. 3-5 hours a day, 2 times a week. Ideal candidate should be positive, patient, energetic. Looking to hire immediately.

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    Місцевий
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    Vicidial webform Закінчився left

    ...group=MORTCP&channel_group=MORTCP&SQLdate=2018-05-22+113857&epoch=1527014338&uniqueid=1527014305.751&customer_zap_channel=SIP/enterprise-000000ff&customer_server_ip=&server_ip=69.64.71.182&SIPexten=gs102&session_id=8600056&phone=7029973001&parked_by=55502&dispo=&dialed_number=7029973001&dialed_label=MANUAL_DIALNOW&source_...

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    Asterisk Management System with PHP Закінчився left

    ...not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create

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    Asterisk dialplan Закінчився left

    ...can be achieved with a call conference or call bridging. when dialing user (A) calls user (B) they are put into a call conference or bridge. if the dialer(user a) hangs up(sip 487) before user b answers, a disconnects but the conference / bridge channel stays open until b hangs up or for a max of 30 seconds cli needs to be passed from a to be etc

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    Moile Sip App coding Закінчився left

    We need sip protocol writer. There is sip application for testing for voice also need recording of voice on app

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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio in chunks to google speech 3. get the recognized text back I would like to either detect the silence to send the audio to speech service or continuous send and get the text as it is ready

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    Install IP PBX Закінчився left

    ...traverse voice over ssl. 3. Ensure that traffic is passed through blocked gateways. 4. Configure auto provisioning for IP Phones and mobile clients. 5. Mobile clients must be compatible across various OS. 6. Configure SIP trunk for International calls (preferably google voice). 7. Any other work that shall be required for smooth implementation and functioning

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