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    8,293 sip webphone tugasan ditemui, harga dalam USD
    Yeastar PBX Configuration 6 hari left
    DISAHKAN

    Require someone familiar with Yeastar MyPBX and S300 setup to configure for WebRTC & SIP

    $148 (Avg Bid)
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    3 bida

    ...Landline Based/VoIP modem based calling. 7. Call forward to Local Mobile or simultaneous ring or International Mobile (Roaming/Holiday) I would need the services to get 1. Dialer/SIP software in iOS/Android. 2. Voip Modem (I have contacts in China to buy the hardware) settings 3. VoIP system deployment on cloud/servers 4. Web Application for customers 5.

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    4 bida

    Hello Our current development website is https://didicar.ca. We need to add Civic plugin to the website. [login to view URL] We tried to install several times, but cannot install and found some php errors. If you have experience with civic plugin, that will be plus. If this project going well, there will be more

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    25 bida
    Trophy icon PAINTnSIP | DESIGN A LOGO 21j 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)
    Dijamin
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    166 penyertaan

    Outbound: -Realtime Monitor - here we can see the agents realtime -Scoreboard - Overview of top 10 agents -Manage Dialer - Here i can give a boost to dialer & and the...here like 77 voicemails 120 not interested -CDR -Batch Report - here i can see how good the imported data is. Ex. data A) 20 sales Data B) 10 sales -Agent Report Settings: -Sip Trunk

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    Twilio SIP set-up 3 hari left
    DISAHKAN

    Please set-up Twilio phone number with SIP so I can use on following site: [login to view URL]

    $19 (Avg Bid)
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    4 bida
    VOIP - CALL DROPS 2 hari left
    DISAHKAN

    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.

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    OBD CALLING 1 hari 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 bida

    ...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 [login to view URL]: Caller ID name is...

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    2 bida

    ...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 [login to view URL]: Caller ID name is...

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    9 bida

    ...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

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    1 bida

    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.

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    1 bida

    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 bida

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

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    4 bida

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

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    7 bida

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

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    6 bida

    ...Prevost, Quebec) Tax Rule rate = 14.98% 2min outbound call on SIP Canuk 200 plan 0.020 = 6sec increment 120sec 2min * 0 .020 * 1.1498 = 0.045992 (round up to 0.045) 3min inbound call on SIP Canuk 200 plan 0.025 = 6sec increment 180sec 3min * 0 .025 * 1.1498 = 0.086235 (round up to 0.086) SIP Canuk 200 Package Detail [login to view URL] & [login to view URL] (already e...

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    9 bida

    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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    50 bida

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

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    4 bida

    Hello, i want script to Test sip accounts with Back SIP response codes [login to view 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 bida

    hello, i have this package : [login to view URL] need to...package : [login to view 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 bida

    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 bida

    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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    11 bida

    I got a business phone system and would like to integrate into salesforce. The system already have API. Requirements: 1) Would like to have webphone(basically a link from our API) into CRM(salesforce) where they can click and call. 2)If can also have recordings(also a link from API)in the crm 3)Have CRM data(name and phone) input into out API

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    37 bida

    I got a business phone system and would like to integrate into hubspot. The system already have API. Requirements: 1) Would like to have webphone(basically a link from our API) into CRM(hubspot) where they can click and call. 2)If can also have recordings(also a link from API)in the crm 3)Have CRM data(name and phone) input into out API

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    4 bida

    hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [login to view 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 [login to view URL] will be long term project if satisfied with the [login to view URL] budget is 100$ for this project

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    13 bida

    Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [login to view URL] /hangupall [login to view URL] /answer [login to view URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [login to view URL] msip:hangupall [login to view URL] msip:answer [login to view URL] msip:38192014 - add new methods that can...

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    hello, please check this doc [login to view 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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    ~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 bida

    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 bida

    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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    2 bida

    ...like for you to do is: 1. Configure a Linux Instance (Please provide the build and we can organise it with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both Media and Signalling

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    9 bida

    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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    3 bida

    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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    9 bida

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

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    6 bida

    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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    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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    25 bida

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

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    10 bida

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

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    8 bida

    ...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 using the "nathelper" module 2. Simply relay the SIP connection to an EXISTING PBX server (3CX). ***Kamailio should relay ALL SIP requests

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    4 bida

    Sila Dafter atau Log masuk untuk melihat butiran.

    Ditampilkan Segera

    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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    22 bida

    ...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

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    Ditampilkan Segera Perjanjian Kerahsiaan
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    2 bida

    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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    6 bida

    ...the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling protocol) with the multimedia framework and NAT traversal functionality. This API is suitable for all types of systems including mobile handsets. Group video chatting

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    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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    18 bida

    ...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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