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    16,174 asterisk pbx jobs found, pricing in USD

    the Telecom project regarding moving the my existing project phone lines to the enterprise PBX and they suggested for us go with MS Teams instead of Avaya. Apparently they are in process of moving admins to Teams (using soft phones) and keeping Avaya as the solution for agents/staff. they suggested using Teams for the office staff as well going to a desk phone (or soft depending on preference). MIgration project to migrate the fax hard phone line to PBX enterprise solution and microsoft teams solution. Need requirements and gathering.

    $255 (Avg Bid)
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    2 bids

    the Telecom project regarding moving the my existing project phone lines to the enterprise PBX and they suggested for us go with MS Teams instead of Avaya. Apparently they are in process of moving admins to Teams (using soft phones) and keeping Avaya as the solution for agents/staff. they suggested using Teams for the office staff as well going to a desk phone (or soft depending on preference). MIgration project to migrate the fax hard phone line to PBX enterprise solution and microsoft teams solution. Need requirements and gathering.

    $10 - $30
    $10 - $30
    0 bids

    Hi I have an ongoing problem from few hours. My server appears to be performing a portscan toward 'random' ip addresses and udp ports. The process that is sending those packets is Asterisk which was already at version 16.29 and I've upgraded it to the last version, 16.30 but the problem persists. I can't provide ssh access to the server but I can provide whatever packets captures you prefer. I'd like to know the source of the problem, if it's some virus in the office of the client or some external attack. Max 100 euros.

    $201 (Avg Bid)
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    setting up an Asterisk server 6 days left
    VERIFIED

    The following is a list of requirements for setting up an Asterisk server: 1. Installation and configuration of the Asterisk software on a Linux server. 2. Interconnection of the server with existing telephony infrastructure (e.g. PSTN or VoIP lines). 3. Setup of user accounts and extensions for making and receiving calls. 4. Configuration of call routing 5. Implementation of call statistics and monitoring tools for tracking usage and call data. 6. Creation of an invoice generation system for billing users for their call usage. 7. Integration with a web-based management interface for easy administration and monitoring of the system. 8. Testing and troubleshooting of the system to ensure proper functionality. 9. Implementation of security measures to protect the system from un...

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    Requirement: We are seeking a skilled developer to create a custom page that integrates with the Brekeke PAL WebSocket. The page should include the following features: 1. Agent login screen where PBX credentials can be entered. 2. Display caller CLI for the agent on the custom page. 3. Fields to capture caller details and store them in a new database table. 4. If the call_type is set to "New_lead", pass the caller's details to a customer application through their API. 5. Display call logs for the current date and allow users to search for call logs by specific date or period. 6. Call logs should include the caller CLI, call timestamp, call remarks, and a click-to-call function. 7. The click-to-call feature should dial out the customer number associated with the call...

    $350 (Avg Bid)
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    18 bids

    Hello I will need a mobile app for android to transfer calls from asterisk to skype (viber) and from skyp(viber) to asterisk...alone audio calls. I will wait your answer.

    $300 (Avg Bid)
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    1 bids

    Hi, I have several asterisk 11 servers. I need: 1. Export inbound and trunks 2. Merge duplicate data. 3. Import data to asterisk 16/20 server. Notes: Code will be reviewed and approved. Code will be tested on tests servers. Developer will not have access to production servers. DM me for any further questions) Thank you.

    $30 - $250
    Featured Sealed
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    Hello Freelancers, I’m looking to implant live DTMF logging into my agents panel using asterisk. The DTMF will be read from asterisk server and will be shown in my agents screen in real time.

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    We need a Linux expert to install faxing in asterisk..

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    Hello all, I would like to capture live DTMF information through asterisk and make it appear on my agent login panel. Essentially all it will do is when a call gets connect and an agent ask the customer to enter a code through their key pad it should appear on the agents panel.

    $187 (Avg Bid)
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    We have an extension which is registered to the asterisk server.. but when making a call the Asterisk returns 401. The task is to find out: 1: why this is happening 2: what we need to do to fix it 3: Make test call to prove fix We will only pay for a fix..

    $27 (Avg Bid)
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    I have an Alpine docker image with Hylafaxplus and iaxmodem and asterisk. my phone provider is twilio. I try to send faxes, but there are a lot of errors. here are some errors. Unspecified failure to train with receiver No response to PPS repeated 3 times. No receiver protocol (T.30 T1 timeout) I imagine that there is a problem in the configuration, can you help me? thank you

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    Hello, I am looking for python experts having good expertise in following areas: Neural Network/ Natural Language Processing Machine learning/Data Mining Deep Learning and Computer Vision Image Recognition & Artificial Intelligence AI text analysis model and Reinforcement Learning. Omnet++ and Sumo simulation Asterisks PBX NS3 simulation Linux I am looking for long term work relationship. New freelancers are warmly welcomed. Important Note: I need dedicated freelancers who strictly follow the deadline and give me good quality work without any plagiarism.

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    What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time ...

    $23 / hr (Avg Bid)
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    Can not send “@” in sms by Dongle Usb Example: I send: test @ I get: test ¡ What im missing in dialplan ? [textmessage] exten => 111,1,NoOp(SMS receiving dialplan invoked) exten => 111,n,NoOp(To ${MESSAGE(to)}) exten => 111,n,NoOp(From ${MESSAGE(from)}) exten => 111,n,NoOp(Body ${MESSAGE(body)}) exten => 111,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) exten => 111,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) exten => 111,n,NoOp(Send status is ${MESSAGE_SEND_STATUS}) exten => 111,n,GotoIf($[“${MESSAGE_SEND_STATUS}” != “SUCCESS”]? sendfailedmsg) exten => 111,n,Hangup()

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    I need to Configure SIP trunk between the billing platform MagnusBilling and the MultiTenant PBX. I need to set it up so that all tenant PBX's are able to call out from the PBX and get billing individually by the MagnusBilling platform.

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    We require someone who can integrate php agi in asterisk box to replicate the similar commands to control dialplan as given under below link here, To enable our customers to run their own IVR using API and SDK Architecture using Cent OS, LAMP framework, Asterisk and PHP-AGI User Accounts connects using API and run their own IVR business logic Our VoIP Arch -> Connected to multiple sip endpoints for each users on their account. I need some one who can help me setup an asterisk on linux machine probably in some better datacenter such as aws or azure that could run behind an proxy public ip for client server connection. And than use php agi to develop an sdk that could send and receive rest api and xml commands for controlling ivr dialplans

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    Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.

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    I have developed an automated system to make calls in asterisk, this is with a , the problem is the calls do not go out and I was investigating, the reason why the calls do not go out is because they do not include exten => _91XXXXXXXXX,1, AGI(agi://). the problem is that I don't know how to include it in the

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    Hi Amal A., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    our company has an asterisk phone pbx connected to local carrier and additional voip carriers. we are migrating all our systems (accounting, sales, etc) under Odoo. We are looking for someone to configure Odoo VOIP module and connect to carriers.

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    I have purchased Acrobits, I need to setup external provisioning to Fusion PBX.

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    Urgent NDA
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    Hi Ghulam Mustafa Z., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    Hi Aleem R., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    Hi Ghulam Mustafa Z., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    Hi Ervin S., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    I need someone to recommend, install and configure OpenSource asterisk based billing system.

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    MIKOPBX Call routing Training We are seeking a Russian or Ukrainian freelancer with specific experience in configuring and working with MikoPBX. The freelancer will provide training. Should have previous experience in Asterisk Dialplan and configure gsm gateway for incoming and outgoing calls to setup for call center this is open source PBX need to work on it

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    Traing for Dialplan and Call Routing for MIKO PBX

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    hi we need call routing training and one login page for extentions to see thier CDR and call recordings that already available in main Admin panel we just need to show them under each user its MIKOPBX opensource based on asterisk

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    Hi I'm not a Cloudflare expert, I have an account with spectrum enterprise enabled (trial) and an Asterisk pbx server with a websocket webphone and stun. I need help to configure Cloudflare in order to proxy the voip calls. Max 100 euros. Please bid only if you have experience with Cloudflare and custom applications (spectrum).

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    Requirement: to be a native speaker of the Turkmenlanguage. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ‚Äč‚Äčrepresented on the record (for Turkmen, you must also indicat...the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Turkmen, then it is necessary to indicate “Turkmen(dialect name)”; c) if in addition to Turkmenthere 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: “Turkmen(dialect), spanish*”; 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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    Deploy a cloud voip solution in aws configure application for multi Tennant pbx deploy sbc and configure and test

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    The goal of this project is to create a step-by-step guide for configuring 3CX (a software-based PBX system) to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown and will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using 3CX or Martini Security's offerings. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to any existing documentation on using 3CX with STIR/SHAKEN that can be discovered on the internet. Martini Security offers a certificate

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    The goal of this project is to create a step-by-step guide for configuring Asterisk/FreePBX to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown and used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Asterisk/FreePBX or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Asterisk/FreePBX with STIR/SHAKEN (https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN). Martini Security

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    We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

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    We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

    $529 (Avg Bid)
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    The Fusion Pbx provide telephony services – Hosted PBX, Toll-Free Number , DID Number/Local Number , Dialer Services , Forwarding Portal , CC Route and VoIP Phone Services .

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    I need a cloud FreePBX system to manage my sip numbers. In the meantime, I need to use database to store all of the CDR (including incoming and outcoming, etc.). Moreover, the instant incoming calls need to show to a webpage on my front-end app, and the response from the app (such as answer, hang up, hold on) will send back to the Asterisk.

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    Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

    $350 (Avg Bid)
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    Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

    $350 (Avg Bid)
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    ...receive speech packets and data. 3. We need to make certain enhancements to the interface and communication between the UniMRCP plugin and the tone / speech detection library we own. 4. This project will require you to work about 12-16 hours a week for 6 months. 5. Skills: (a) Pro in C/C++ (b) Familiar with UniMRCP plugin development architecture. (c) Telecommunications Engineering, Asterisk, Socket Programming, System Programming IVR Software, gRPC, Websockets. 6. Deliverables (a) Fully-functional plugin in executable form along with libs. (b) Product must be containerized. (c) Complete source code of interface. (d) No GPL code to be used (e) Documentation: (a) User Documentation or README file to setup/run, (b) Design and source code documentation. ##Yo...

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    We experiencing the issues with audio in IVRs and MOH under high load(300-400 concurrent calls). They are breaking up and very choppy. The CPU load with 400 calls is around 50-60%. There is no issues with about 150 calls. The server details are below: CPU: Dual Xeon Silver 4309Y (16x2.8GHz) RAM: 32 GB SSD: 1TB(RAID 1) Asterisk: 16.21.1 Front End: Issabel It should be enough resources to handle this load. We need someone who can analyze the issue and provide the solution for it.

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    ...use the available DTMF commands already available for routine conference calls. We have the technical documentation and source file samples available, these code samples are based on Asterisk, and they give us an overall view of how to develop the adaptor and allow to interconnect the Audio Reservationless Bridge/Platform to Adobe Connect. The Audio Reservationless Bridge/Platform already working and has been tested. This API/Interface/Adapter will be considered IP and owned by us. See the attached drawing for additional details. We are seeking an experienced developer intimately familiar with Asterisk. You must beable to start on the project ASAP. Thanks Tom! ...

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    Build UP PBX server + Billing ( user , admin , etc)

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    Hello, goal: make BLF (Busy Lamp Field / Presence) with VitalPBX (Asterisk) work for Linphone. Let's discuss via chat. Thanks, Christian

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    This project is about customising Vicidial & Asterisk applications. + 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 have the list with a multiple choice test. The problem is that the right answer is always placed first among the three possible answers. Furthermore, the answers are numbered and the right answer has an asterisk, as well as always being the first one, the right answer. I would like that the right answer be randomized among the three of every single group, to eliminate the numbering 1, 2, 3 and remove the asterisk. You can give me the new file in pdf, word. Google sheets or excel. You should not touch the questions. I attach the file. Thanks.

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    Hello, I would like you Caller Id asterisk sip that can be used anywhere in the world. Thank you

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    We use 450gx4 for our office with dual wan with failover and load balancing. We would like to do the following 1. The load balancing and failover should work perfectly 2. Only one wan has real ip I need to access the ip. Pbx through that ip via ssh and Web the mikrotik remotely on the real ip vpn server 5. Bandwidth management for users so that they can use fairly graph for both wan and bandwidth usage for. User report

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