Asterisk pbx工作

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    16,110 asterisk pbx 份搜到的工作,货币单位为 HKD

    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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    anyone how has experience with making cisco 6901 phone to work on free PBX SCCP manager

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    We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams

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    We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams The goal of this project is: Configure an Opensips server with Opensips Control Panel that: - Connect to asterisk PBX server / Fusion PBX - Connect to Microsoft Teams - Let users from MS teams call users on Asterisk / Fusion Extensions and external calls though this PBX - Let users from Asterisk/Fusion call users on MS Teams

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    I currently use a variation of salesforce as my crm and FreePBX as my pbx. I would like to setup some zaps in my zapier to log each call made with date, time, and link to recording. I want this to occur on both inbound and outbound calls. If no record is found then I do not want one created

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    Tenemos una pbx issabel, y queremos tener un interfaz más amigable con funciones de centralita virtual, y que este enlazada con nuestra pbx issabel. Se valorará trabajos similares y nos enseñe ejemplos

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    Hello. I have 125 Google sheets of Multiple-Choice Question Data. These are all formatted identically and an asterisk is used to identify the correct answer. An example of the first sheet of data can be seen here. To import all of these quizzes into my website I need to reformat each sheet automatically to match the new template. The biggest change here is that an asterisk is no longer used to identify the correct answer and instead it is a numerical value of 1-4 which indicates if the correct answer was A,B,C, or D. Ideally, I would like to be able to copy the original sheet of data into a book, run a script and end up with the data formatted correctly for export. An example of the new

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    I have a white-labeled Linphone application for both iOS and Android completed and use Asterisk servers. Push notification for incoming calls is causing me challenges. I'm looking for someone to hire to: Walk me through setting up my Apple developer account and Firebase to send push notifications to my Linphone build. Walk me through associated modifications to Linphone build (integrate Google plist, etc, whatever needs doing). Configure kamailio (preferred) or flexisip (acceptable) to proxy between my Asterisk systems and Linphone on client mobile devices to handle push. We'll use a fresh Debian 11 install that I will provide you credentials for. If this is something that you can take on, please provide a quotation.

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    Asterisk + Freeside (billing + Self service) installation Debian 10 VoIP service setup and installation on a Debian 10 server

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    I have a white-labeled Linphone application for both iOS and Android completed and Asterisk servers. Push notification is causing me challenges. I'm hoping I can hire you to: Walk me through setting up my Apple developer account and Firebase. Walk me through associated modifications to linphone build. Configure flexisip proxy to handle push on a fresh Debian install that I will provide you credentials for. If this is something that you can take on, please provide a quotation.

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    requiero un manual o tutorial para implementar seguridad en el servidor asterisk, ya sea con fail2ban, iptables. para mitigar los regitros sip, ssh y los ataques de denegacion de servicios. Acepto cualquier sugerencia para mejorar dicha seguridad. I require a manual or tutorial to implement security in the asterisk server, either with fail2ban, iptables. to reduce sip, ssh logs and denial of service attacks. I accept any suggestion to improve said security

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    Trophy icon Logo for Cloud PBX 4 小时 left

    Hi guys, I need to create a logo for a new PBX project. The logo must be delivered in at least these formats : Vector AI, EPS, JPEG, PDF, SVG PNG, PSD. We need a lifetime guarantee for the logo and the changes in the future must be completely free. The logo need to contain the word "ILPBX". The logo must be previewed in colour, in relief, in contrast on a white background and on a black background. The writing "ILPBX" must also be placed as an alternative to "". The message that the logo will have to send to the viewer is something that suggests business telephony, smart communication between various devices such as smartphones, landlines, PCs and cordless phones. However, it must be simple and clean.

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    WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk

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    Im looking for someone to build be a self hosted voip system We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc'

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    I have the CRM link module installed on my pbx and conected with my crm (Improvit360 /Salesforce) I need the module modified to check the numbers against "prospects" and "customers" in the CRM and add the task from the call to their records. (Normally the CRM Module checks for accounts, leads, and contacts. I do not use these) I also need the call recording attached to this record as well.

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    We need to modify just one function in open-source software that is connected to our Asterisk (FreePBX) server. We know where exactly this change should be applied. So it would be very straightforward for a java experienced developer. Asterisk, AGI, AMI, Java-asterisk lib, Java, J2EE, PHP, Python

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    voip system 已经结束 left

    Hello We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc' The end goal is to be able to start selling voip services. We prefer a person who did something like that before. I will be happy to answer questions.

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    Asterisk troubleshooting 已经结束 left

    Hi - we have an asterisk box V16 running on Centos 7. We're having ongoing issues with CPU spiking with as little as 3-4 calls, reboot of the server temporarily resolves the issue but we need to resolve the root cause.

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    Asterisk Server Script not working I will send details in chat

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    The CRM I use is Improvit360. This is Salesforce branded for the home improvement industry. I need my PBX (FreePBX) to be integrated with it. I have the module on Freepbx to connect them but it needs to be setup through REST API.

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    Hi, We need someone who can upgrade our FreeSWITCH and OpenSIPs to the newest stable versions on Amazon AWS. Currently we use FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is...2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is 1.10.8 We also need OpenSIPs upgraded to the newest version 3.3.2 we currently are on: 3.0.2 (x86_64/linux) This is a live production server so it will need to be done pretty quick in a couple hours or so. If we work well together I will have many more ongoing tasks involving FreeSWITCH, OpenSIPs, our PBX and other issues, our main telecom engineer/developer was in Ukraine and we have not heard back form him in months. Thank you! Tha...

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    I have the CRM link module installed on my pbx and conected with my crm (Improvit360 /Salesforce) I need the module modified to check the numbers against "prospects" and "customers" in the CRM and add the task from the call to their records. (Normally the CRM Module checks for accounts, leads, and contacts. I do not use these) I also need the call recording attached to this record as well.

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    small fix on current astersk pbx which is running on dedicated linux server. Not receiving calls and problems with outbound calls. Only experienced

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    Modify an script 已经结束 left

    We need to modify just one function in open-source software that is connected to our Asterisk (FreePBX) server. We know where exactly this change should be applied. So it would be very straightforward for a java experienced developer. Asterisk, AGI, AMI, Java-asterisk lib, Java, J2EE, PHP, Python

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    Unity Developer 已经结束 left

    ...Ambient music + creature noises 10. Effective draw distance for a multiplayer web browser game 11. Proximity voice chat (based on player vicinity to sound source or other player) 12. Stream a real-time video feed (2D Sprite) of Maghmul into the Unity environment* 13. Movement mechanics (WASD/Gamepad + Jump) for players and real-time videostream* 14. WebGL Deployment and testing * Items with an asterisk are already in progress or completed Kindly refer to the videos linked at the bottom of this document for more information on the status of the project Ideal Candidate 1. Would be able to provide fixed-bid quote or estimate for the items listed in Phase 1 following a 4hr introduction to the code base. 2. Have 3-5 years full-time experience developing games or experiences in Unity ...

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    Create custom module in prefex crm for integration with asterisk with following features: - create/edit extensions - create/edit trunks - create/edit outbound routes - create/edit inbound routes - create/edit IVR menu - calls report and recording - popup up screen when incoming calls with customer details or add new option. - click to call from crm. - conversation history in customer details. Thanks

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    Project Requirement: 1) Design one Linux Script with netcat (nc) or standard C program for sending the "Caller ID" from my Asterisk IP-PBX to the WordPress Web server. The Linux Script Command options will look like "send_hostip_callerid". 2) The WebPress Web server need one simple Server Daemon to receive the "Caller ID" sent from my Asterisk IP-PBX by "send_hostip_callerid". The Server Daemon on WordPress Web Server need to support multiple concurrent incoming connections or you can use the CGI program of the WordPress Web Server to handle this job. After this daemon or CGI program has received the "Caller ID" from "send_hostip_callerid" on Asterisk PBX, it must save the "Caller ...

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    + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

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    I am setting up our FreePBX host and need help with the firewall rules for pfsense, ensuring our windows server (AD/DS) are set up correctly to allow secure port forwarding. Mainly, I am on the portion of setting up a let's encrypt certificate on the FreePBX device and am unable to make this portion work successfully.

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    Modify Freepbx CRM Module 已经结束 left

    I have the CRM link module installed on my pbx and conected with my crm (Improvit360 /Salesforce) I need the module modified to check the numbers against "prospects" and "customers" in the CRM and add the task from the call to their records. (Normally the CRM Module checks for accounts, leads, and contacts. I do not use these) I also need the call recording attached to this record as well.

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    My end goal is to have a FreePBX (asterisk- AWS instances) and yeastar ippbxs (asterisk but physical device) to push CDRs and recordings to a central panel which the end user will get access to. the yeastar devices don't have fixed IP dresses but we can use DDNS to bypass that issue from the outside network. also can use AMI. the FreePBXs do possess fixed IP addresses, so in that matter we won't have issues. both of them suppose to send cdr data and recordings to an s3 bucket which will push the recordings to a panel which will be built for the end user. ( we can push all data to be stored at S3 and from there to pull it with the panel) the panel suppose to be separated per user so for example, user is called Adam and he has 3 workers, Only Adam will have access ...

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    Customer installation consists of a 3CX PBX, Patton T1 VoIP gateway with 2 T1 connections for a total of 46 channels. Customer is experiencing that some incoming calls drop after a few seconds and 3CX reports "route busy". Have sent Wireshark captures to 3CX and Patton tech support and unable to resolve the issue. The problem could be the T1 provider. Need a network engineer to work remotely capturing SIP data and determine the cause of the problem.

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    ...ability to plan, implement or oversee coordinated testing of new applications, to ensure compatibility with existing hardware and software applications. · Responsible for engineering and/or analytical tasks and activities associated with areas within the telecommunications function (e.g. engineering, implementation, diagnostics or operations/user support). · Monitors the operation of telecom (PBX/VOIP) systems. · Performs complex tasks relating to telecom operations, installation, and/or maintenance for local, off-site and/or remote locations. The scope of responsibility for this position includes, but is not limited to, the configuration, deployment, testing, maintenance, monitoring and troubleshooting of telecommunications network components to provide a...

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    -Install SMS manager server & configure it with GOIP gateway - sms config in web freepbx install configure extensions -config name number sim in portal and pbx income

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    Looking for a server admin with experience in Windows Hyper-v / TrueNAS / SIP / SBC / PBX to help with management and setup for our server cluster as well as voice customers. Skills in networking including routing, BGP, VPLS, Mikrotik etc would be an added bonus. This would be a full time position covering overnight USA time zone.

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    Looking for a server admin with experience in Windows Hyper-v / TrueNAS / SIP / SBC / PBX to help with management and setup for our server cluster as well as voice customers.

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    Arduino SIP client 已经结束 left

    Need a Arduino SIP Client for getting connected with asterisk pbx

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    ...self-hosted freepbx 16 server but we are having issues with calls. Sometimes we can't be able to hear other person but they can be able to hear us. This is randomly happening with all the extensions. After multiple times try calling again and we can be able to hear each other. This is happening on both inbound and outbound, peer to peer and external calls. Below are the repeated errors on the asterisk console ``ERROR[30743]: res_rtp_asterisk.c:4881 ast_rtcp_write: RTCP SR transmission error to client_ip:40011, rtcp halted Operation not permitted`` `` ERROR[30743]: chan_sip.c:4354 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data`` We need an expert to fix all the issues with our freepbx system and make sure to resolve all the calling iss...

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    Hello i got a Yeastar S20 and i have MicroTek i need to configure to allow me to pass the blocked ports in my country also i have another one in another country but it's works well ,, any experts ?

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    + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

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    Features needed Account Management Call Rates Call Routing Strategies Calling Cards DID Management Invoicing And Billing Payment Gateway Product Management Rate Groups / Tariffs Reports And Alerts Reseller Account System Settings Multi Tenant IP PBX

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    I'd like to setup a multi tenant asterisk VoIP system with WebRTC. I also want two simple clients that can all call each other. These should be: - A simple WebRTC web client - A simple iOS WebRTC client built with Swift with built-in CallKit and VoIP push notification integration The purpose of this project is to build a proof of concept project to test making calls. The system should be built with best practices and with scalability in mind, but this won't be a "finished" product. We can discuss the details in a call if interested.

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    I am looking for someone to make a simple call flow in bound dial plan for asterisk, should be as follows . call to did - dial plan will send the call to whichever agent that is free and logged in sip account

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    + Ideally Asterisk Coding experience is desirable / MUST + You must have very good experience with Customising Asterisk / Vicidial according to requirements + Several Asterisk / vicidial Customizations should have been completed + Experience with Create custom call flows, Dynamic agent assignments, TTS, Speech to text is required

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    I need you to implement Asterisk telephone server on Raspberry

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    Hi Kamal. I'd like to setup a multi tenant asterisk VoIP system with WebRTC plus simple WebRTC web clients. Can you help with this? We can discuss the details in a call if interested.

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    Develop a new pbx system voip with modern interface (need to have images to see) where we use didww for the numbers which are ready and setup call routing, music on hold, setup call conference and schedule and rest is all normal routing and dialing. Call flow is available. Freepbx or vicidial or any other system that is modern and easy to setup as it has to be ready end of the week.

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    i would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in to it use a SIP software in order to answer customer calls. What I want is a convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field and last three data pulled out to select to go into the dispatch panel Deploy PBX Create few extensions Create script for take CID of inbound calls check last 3 records, sent CID and last 3 records via api to dispatch app Help to setup tranks and dialplan

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    Need to shift Issabel (Asterisk) live recording data store to AWS S3 Storage. Please check the attached file to know exact work needed.

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    Hey, Looking for someone to create Call History List from FREEPBX (Based on Asterisk) by their API using PHP. Instant Work.

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