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    15,969 asterisk pbx 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

    $55 / hr (Avg Bid)
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    1 入札

    Hi i have installed asterisk on server , want to set sip dialer (soft phone for voice calling) my admin portal is on laravel platform, i want to set phone on that , which will be connected to asterisk on backend, right now m using zoiper , but i want to have web app. SIP VOIP Asterisk VPN

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    Hello We need to build auto dialer + IVR Survey call on asterisk After this we need to integrate this PBX with ERPNext CRM Module

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    3 入札
    PBX + chat 4 日 left
    認証完了

    Hello I need server configuration for PBX include chat, you should have previous experience in server configuration. our application is only android and you should build the server side. project requrments is: 1. internal voice call 2. chat 3. group chat i am only working with exprienced developer in this field

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    $26 平均入札額
    3 入札

    Hello Guys, We are looking for someone who can help us to setup a multi tenant freePBX to start a basic Toll Free Number providing services to local businesses.

    $228 (Avg Bid)
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    8 入札

    Hello, We're currently seeking a service provider with previous experience in Flutter or React Native to build an application for multiple platforms like IOS & Android. The candidate should have previous experience in building communication applications and also have a strong understanding of how VoIP, PBX, Asterisk work & interact behind the scenes. The app will be integrated with the open-source system, Asterix. The goal of the project is to replace the existing 3rd party solution we're currently using called Zoiper. If you have experienced in building communication apps and understand Asterix, VoIP, FreePBX & Flutter, please reach out for more information. We have AdobeXD designs & technical specifications document with use cases wh...

    $7091 (Avg Bid)
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    81 入札

    Have a Granstream UCM6208 New PBX . He needs help configuring it for two landlines and two phone extensions. We are creating Group Intercome. We could create a remote session. Please provide a price quote. Also, I need documentation on how to add additional extensions when required.

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    7 入札
    OpenSIPS B2BUA 2 日 left

    We need assistance in setting up OpenSIPS as a B2BUA between a PBX and an SBC. The B2B will also need to manipulate the SIP Header info on egress.

    $616 (Avg Bid)
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    FreePBX Firewall Configuration 1 日 left
    認証完了

    My FreePBX Server is having issues with call deployment due to the firewall and fail2ban. I need someone who has hands on knowledge of configuring it all way to make it work. I am myself a Full Stack Developer and have good knowledge, so people who really knows what Asterisk SIP Settings and FreePBX Firewall is actually apply.

    $257 (Avg Bid)
    NDA
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    VOIP project 4 日 left

    Hi Eremin P., I noticed your profile and would like to offer you my VOIP project. Cloud PBX based on Opensips and FreeSWITCH. This would be a long term project. We can discuss any details over chat.

    $20 / hr (Avg Bid)
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    1 入札

    ...for development of a Zimbra Collaboration Suite add-in (called a "zimlet" (java)) which can talk to Asterisk (soft-PBX) using Asterisk AMI to allow calls to be initiated (the call is started and then your internal deskphone starts ringing and is connected to the outgoing call. Both Zimbra and Asterisk are open source Linux based products. There have been open source projects to do this but based on older versions of ZCS and none support ZCS 8/9 and also Synacor even include click-to-dial for specific phone systems but not Asterisk. Here is a youtube video that explains how to create zimlets for ZCS 9 Here is a youtube video that explains how the Asterisk AMI can be used: Here is more documentation about zimlets

    $687 (Avg Bid)
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    ASTPP & FusionPBX expert 6 時間 left
    認証完了

    Hello, I am considering to adopt new deployment using ASTPP for billing to be integrated with FusionPBX utilizing the multi tenant IPPBX typically ASTPP will handle the billing aspects for voip termination and DIDs origination (we currently use IDT Express and ) then map it conveniently to the PBX tenant I will surely avail both cloud VMs with the base OS as per you recommendations most likely will be Debian as per the documentation or of you want me to setup boh ASTPP and FusionPBX beforehand that would be ok too and then you will just help me onboarding with the initial setup, configuration, interconnection and integration between both VMs and between ASTPP with both and IDT to be acquainted with the full picture and end to end cycle. please note that I already

    $246 (Avg Bid)
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    5 入札

    Need to improve certain area of the server as following. Server details: Current VoIP system: FreePBX (Asterisk 16.22.0) OS: Ubuntu 18.04 Database: MariaDB 10.4.22 Tasks: - Enable and configure Ubuntu firewall UFW - Install and configure Fail2Ban. When BruteForce comes on SIP registration try, register bad IP addresses in the list for 3 days to block any access. - Enable MariaDB to listen on TCP 3306. Currently working via ODBC with Asterisk - Install and configure NFS server - Setup MariaDB “Master-Slave” replication - Monitor Asterisk service running. If stops, send email alert and try to restart asterisk once. If you read the requirement, please include word "peace" in your proposal. Thank you.

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    We would like to setup a Asterisk callcentre with suitecrm. . The asterisk needs to be integrated with suitecrm with callpopup , call logs and other standard features. For the CRM it needs to work with messenger ,email , whatsapp We won't like to use any 3rd party addons that we need to pay yearly or monthly. Only offer if you can build the connections yourself.

    $222 (Avg Bid)
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    5 入札

    Hi I moved one Fusion PBX server to another. The destination server crashes every few days or less and does not come back without a restart. And it is not possible to connect via ssh when the server is down. And in my opinion, this is a problem in Linux and not in the aforementioned application. The operating system is CentOS 7 If you know the problem and know how to solve it - please contact me. I am attaching a picture of the console when the server is down

    $75 (Avg Bid)
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    2 入札

    Hello We have the latest FreePBX server set with asterisk in it and Node Red to control the asterisk dial plan. Requirement is when we get an incoming call through a channel then Node Red identifies and takes over the call control. While Call forward and pre-recorded messages works fine however we are not able to get TTS based MP3 audio playback done via node red in asterisk. As per asterisk MPG123 will work however as using by nodered thus the file as to be injected via a function instead of node. Looking forward for solution to make following happen 1. Play mp3 file in asterisk(freePBX) via node red over an extension in incoming call. In a way node red will become dynamic IVR for bridge connections 2. Based on caller ID Node Red will HTTP request our...

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    If you know what you are doing this shouldnt take longer than 20 or 30 minutes... I do not know what I am doing so I need some help. I am not sure what goes where and the terminology confuses me a bit and I do not want to spend time trying to figure this out. Here are in the instructions: Let me know if you can help

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    Hey i am trying to build an app in asterisk. I am facing some problems with configuring file. Need guidance in configuring that file and need some answers of asterisks questions.

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    I am looking for someone who understands asterisk systems or VoIP systems. I need someone who can help support current clients.

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    Looking for someone who understand Avaya PBX configuration with SIP trunking

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    9 入札

    Create an api in asterisks softswitch to handle calling on android and IOS linphone apps The call will work via Local number that is instead of the app calling via internet VoIP. , it will initiate the call via the app and the app fetches the Local number from the astericks softswitch. So when caller hits the call button the api fetches the DID local number from asteriks softswitch then places the call to the destination this works silently and there will be no tap tap sound . And if there is no internet in the phone it will send the call via thesame Local DID number in the astercks softswitch like this Country code,, destination number,,# that is in a pinless calling manner. But this way will make a tap tap sound . There are many apps that do calling using this type of l...

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    I need to develop a cloud call center by using Asterisk in backend and need to create extensions , soft phones and configure from browser, if any one have an idea about this ping me ASAP

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    6 入札

    Looking for someone who understand Avaya PBX configuration with SIP trunking

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    6 入札

    must be set on a raspberry pi 4 asterisk and freepbx with 4 dongle channels. To create dialplan to send automatic calls from cli

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    7 入札

    ...management/monitoring tools, operating systems services (DHCP, DNS, NTP, SNMP, SMTP, BGP, OSPF, VPN , etc ...) - Cloud technologies knowledge(Amazon AWS, SoftLayer anIBM Company, Oracle Cloud, OVH, SoYouStart, DigitalOcean, etc ...); - web applications technologies (Tomcat, Apache, NGINX, Redis, PHP - fpm, Memcache, etc...); - basic telecommunications concepts and protocols knowledge(SIP/RTP) ( Asterisk ) - version control systems (GIT, SVN, CVS) - database engines administration and optimization ( MySQL, MongoDB, PostGreSql, etc ...) - database administration application / installation / troubleshooting / support (Oracle 9i - 12g) - good knowledge of Windows Server (Hyper-V, Active Directory, Group Policy DNS, DHCP, File Server) - experience providing support in medium to large ...

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    I have a trouble install Kazoo on centos Any one who have experience in Kazoo Welcome!

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    8 入札

    -SIP-based call to Softphone (web app) - Video & Audio SIP Integration We want users to be able to join active conference by dialing a specific URI and be able to actively participate using audio and videos and be able to share screen with the sip phone or terminal emulator respectively - Dial Out We want our users to be able to call a user ...meeting using our application (internal extentation or any mobile number) - User to User Call We want our users to be able to make user-user call on our existing mobile app using internet which will support video and audio with clear voice and lower bandwidth. - User to Phone Call We want our users to be users which are not on our APP Features : Recording, Call Forwarding *Have experience in Asterisk Issabel json API If have any solution...

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    13 入札

    Necesito realizar el siguiente trabajo 1.- Configurar una ruta entrante desde asterisk (puro) para que pueda ser enrrutado a una cola y contexto 2.- El agente (a discresión) cuando entre (login) le caigan las llamadas de esa cola o contexto

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    We need a ZOHO CRM & AMAZON Connect expert to integrate the SMS/MMS to our ZOHO CRM so we can see the interaction in the contacts profiles. Please write ZOHO CONNECT on top of your bid or your BID WILL BE REJECTED

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    Hi, We are a startup and need to hire a FreeSWITCH / OpenSIPs telecom engineer to help us with tasks from time to time. We would like to work long term with only 1 developer / engineer, you must also know how to install and setup FreeSWITCH/OpenSIPs on AWS. We will pay by the hour, please send your resume or experience and price per hour you charge. thank you.

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    17 入札

    Need a inbound call centre. You need to set up asterisk server for below features, but then another coder I work with will create a custom panel, and need to integrate both together. So it is a joint project. You will need to create API to integrate with custom panel. Ability to transfer calls to a closer Ability to create agents, 2 types of accounts, closer/opener Ability to add scripts, script will show according to Campaign/Agent type. Example, Car Campaign, show Opener script to Opener agent for Car campaign. Ability to have custom hold music, customise it per campaign or have a default one for all. Agent Alert sound for inbound calls. Ability to have a queue, many people call, all agents busy, play hold music and custom speech, saying 'You are now number 4 in the queue&#...

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    36 入札

    I need a Softphone, developed in .NET which can connect to my Issabel PBX system Need the code to be connected and compiled to an accual .NET development I have

    $2537 (Avg Bid)
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    15 入札

    ...management/monitoring tools, operating systems services (DHCP, DNS, NTP, SNMP, SMTP, BGP, OSPF, VPN , etc ...) - Cloud technologies knowledge(Amazon AWS, SoftLayer anIBM Company, Oracle Cloud, OVH, SoYouStart, DigitalOcean, etc ...); - web applications technologies (Tomcat, Apache, NGINX, Redis, PHP - fpm, Memcache, etc...); - basic telecommunications concepts and protocols knowledge(SIP/RTP) ( Asterisk ) - version control systems (GIT, SVN, CVS) - database engines administration and optimization ( MySQL, MongoDB, PostGreSql, etc ...) - database administration application / installation / troubleshooting / support (Oracle 9i - 12g) - good knowledge of Windows Server (Hyper-V, Active Directory, Group Policy DNS, DHCP, File Server) - experience providing support in medium to large ...

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    40 入札

    hi i want someone to set sip dialer(client - web phone) in front end made in laravel for making audio calls from asterisk server there are so many projects on github for web sip dialer u can set this with laravel. so on clicking on dialer button in front end dialer will open get registered on asterisk server and we can call. he has do set up on server also if required regarding socket for web phone settings are authenticated in database, everything is dynamic voip

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    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Chinese, you must also indicate the dialect)...following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Chinese, then it is necessary to indicate “Chinese (dialect name)”; c) if in addition to Tajik there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Chinese (dialect), japanese*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    19 入札

    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Tajik, you must also indicate the dialect) a...to the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Tajik, then it is necessary to indicate “Tajik (dialect name)”; c) if in addition to Tajik there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Tajik (dialect), turkish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    14 入札

    Each files duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Azerbaijani, you must also indicate t...rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Azerbaijani, then it is necessary to indicate “Azerbaijani (dialect name)”; c) if in addition to Azerbaijani there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Azerbaijani (dialect), turkish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    8 入札

    Hi I migrated one fusion pbx server to another. The target server crashes every few days or less and doesn't come back up without a restart. And there is no possibility to connect via ssh when the server is down. If you know the problem and know how to solve it - contact me please. I am attaching a picture of the console when the server is down

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    6 入札

    we have installed Kamailio with some asterisk server behind and we need some help to configure kamailio to be a loadbalancer and acting as SIP proxy to just pass all sip messages including register to asterisk servers behind we need this to be done asap

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    14 入札

    want customise own call centre interface with agent login free seat. reporting.

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    12 入札

    We have Cisco phone CP-8861 and we want to connect it with Asterisk

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    9 入札

    We need vTiger Asterisk connector expert to connect asterisk and vTiger CRM. Asterisk and FreePBX are on AWS cloud. vTiger is on another server and the edition of vTiger is opensource You must have experience in connecting asterisk and vTiger , please give us your previous work demo link . We have 25 -75$ budget. Keep this in your mind and bid please accordingly Thank you

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    i need to configure asterisk with webrtc

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    7 入札

    ...using VoIP. You're a full stack developer. Requirements: - Backend: FastAPI / Python + Asterisk - Frontend: Svelte / Bootstrap 5 / HTML5 / CSS3 You will: 1. Develop the front-end by taking the mockups into account. 2. Develop the backend & the API routes & all mechanisms to make the front-end dynamic (except the VoIP related features). 3. Install & configure Asterisk with a Telnyx account to be able to do spoofing and 100 simultaneous calls. To test the numbers, automatic calls will have to be done. You must answer the some questions. 1. What is your expertise level in VoIP web apps ? An example ? 2. How will you interface the front-end receiving & sending the audio to the python backend and Asterisk ? 3. Could you provide an example of a Fa...

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    Hi I need a dial plan for an extension in fusion pbx where I will have an extension as follows. I call the did number that points to this extension - the system tells me to dial a phone number for identification (that is, what will be the callerid for the call - what identification will appear at the destination) - the system repeats the number I keyed and gives these options to confirm press 1 to press again press 2 - If you press 1 - please enter the phone number you want to call (destination). The system repeats the keystrokes and so on - if you press 1, the system dials using a trunk that you will have to set up (of my provider) - the call is automatically recorded. And I can receive the recording by email/ see and hear it in Fusion's interface. The budget for this is $80

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    6 入札

    Need someone to build Asterisk Docker image

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    12 入札

    we are required a Asterisk developer to customized the asterisk software and develop the Asterisk based pbx

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    12 入札

    Hi I need help with various settings on the subject of voip

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    10 入札