Freepbx asterisk jobs
...skills - Attention to detail for accurate data entry and call tracking Project Description: We are looking for a freelancer who can assist us with our Vicidial short duration project. The desired duration of each call is less than 1 minute, and the purpose of the calls is sales. We require someone with experience in using Vicidial or similar call center software. This project can be done on asterisk of vicidial web interface as well. Itshould be just a code on the dialplan. The specific features for the dialer have not been specified by the client. Therefore, we are open to suggestions and recommendations from the freelancer. The ideal candidate should have a strong background in sales, with proficiency in sales techniques and strategies. They should be able to ha...
I have a VOIP PBX based on Freepbx . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is engaged and in no other case.
I have a VOIP PBX based on Freepbx . I need to do the following configurations: 1) When a call comes in, both phones must ring; 2) If the call is answered from one of the two phones and in the meantime another call comes in, the caller must simply get the busy signal 3) While talking from one of the two telephones it must be possible to make a call from the other, i.e., the one that is not busy. 4) When a call comes in, and no one can answer, after a certain number of rings (8) play message "operator not available try again later". 5) The answering machine should answer only when it is engaged and in no other case.
I am looking for a freelancer who can help me deploy Asterisk 16 with a specific PJSIP configuration. The ideal candidate should have experience with Asterisk and PJSIP. Requirements: - Familiarity with Asterisk 16 - Ability to configure PJSIP according to specific requirements - Experience in handling concurrent calls, with a focus on optimizing for a single call The requirements are very simple. I have configured it myself before and achieved single-pass. If the extension calls the mobile phone, the sound of the mobile phone can be heard, but the sound of the extension cannot be heard by the mobile phone. You only need to configure the phone to be able to achieve dual communication! If the price is not suitable, the price can be negotiated as long as you can solve...
Project Description: Troubleshoot and resolve registration issues with PBX sip trunk - I am using an Asterisk PBX system and attempting to register a SIP trunk with a Telecom Provider - I am not sure if there are any error messages being displayed when attempting to register the SIP trunk - I am seeking a skilled professional who can help me troubleshoot and resolve any registration issues with the PBX sip trunk - The ideal freelancer for this project should have experience with Asterisk PBX systems and SIP trunk configuration - Knowledge of Telecom Providers and their registration processes would be beneficial
IVR PBX VICIDIAL INSTALLER I am looking for a skilled freelancer to provide a full installation of an IVR PBX VICIDIAL solution. Specific f...VICIDIAL solution. Specific features that need to be supported include call recording, interactive voice response (IVR), and voice mail. The system will be used by a specific number of users, with the exact number to be determined. Ideal skills and experience for this project include: - Extensive knowledge and experience with IVR PBX solutions, specifically VICIDIAL - Familiarity with Asterisk and FreePBX - Ability to configure and set up call recording, IVR, and voice mail features - Previous experience working with systems supporting multiple users If you have the necessary skills and experience, I would love to discuss this...
I am looking for a freelancer to build the latest version of FreePBX system with A2billing. It will be preferable to have it in a docker, although not necessary. I understand that A2billing is no longer supported; therefore the candidate can install and configure any open source calling card platform that can cover most features of A2billing. Thank you.
...dont have Asterisk and SCAIP experience. I am in need of an English speaking Asterisk & Kamailio specialist to assist with my project. The specific tasks that I require assistance with include configuring the Asterisk and Kamailio servers and the settings. Specifically we want help in complying with SCAIP (Social Care Alarm Internet Protocol). If you have this experience that is perfect. We already have an Asterisk engineer but he may need some help with interpretation. I already have existing infrastructure in place, so the specialist will be working with that instead of setting up everything from scratch. The expected turnaround time for this project is moderate, ranging from 1 to 2 weeks. Ideal Skills and Experience: - Strong knowledge and experi...
I am looking for a freelancer who is experienced in FreePBX and can help me solve a problem with my caller ID. Currently, my caller ID is not displaying correctly. Skills and experience required: - Strong knowledge and experience with FreePBX - Expertise in troubleshooting and resolving caller ID issues - Familiarity with PBX systems and call routing - Ability to analyze and diagnose caller ID problems - Knowledge of SIP and VoIP protocols The ideal candidate will have a proven track record of successfully resolving caller ID issues in FreePBX systems. They should be able to quickly identify the cause of the problem and implement the necessary fixes to ensure that the caller ID is displayed correctly. Good communication skills and the ability to explain technical co...
...though it does not necessarily have to be an animal; and we would like to place it in the triangle above the product name. We are looking for an organic font for the product name and something standard or type-faced for the tagline. The profile of the sacha inchi seed (the pulverized version being sold) would have preferred placement between "Ancient Seed" and "Modern Life" as if it were an asterisk separating the two phrases, or an abstracted version of it above the product name is an idea. In regard to color we have leaned toward the red with white boarder, and black, blue, white, and grey around images and fonts. We do, however, like the contrast of the logo against the triangle. That is a start, we are open to comment, question and collaboration. O...
We need our asterisk to play a small audio file to destination at the moment we receive 180 or 183 from destination. Destination provider needs 5 RTP packets to discover IP and port when asterisk is behind NAT.
I am looking for a freelancer who can create a custom context for my FreePBX system. The ideal candidate should have experience with FreePBX and be knowledgeable in creating IVR menus (the bid need to be the real bid of your cost for make it please) Specific Features: - I need the custom context to have IVR menus Integration: - The custom context should be integrated with my CRM system - The IVR should be able to capture data dialed and save it into a database Data Storage: - The captured data should be stored in the database as individual fields If you have the necessary skills and experience, please submit your proposal. (attached it is a document with the workflow)
I am looking for a freelancer who can assist me with data entry work. Specifically, I need help with typing and data input. I already have the data that needs to be entered, so there is no need to worry about finding the data source. The volume of data is ...about finding the data source. The volume of data is relatively small, with less than 100 entries to be entered. Ideal skills and experience for this project include: - Proficiency in typing and data entry - Attention to detail to ensure accurate data input - Ability to work efficiently and meet deadlines - Familiarity with data management and organization The correct answers have an asterisk next to them just need to copy and paste across essentially I am up to question 17 follow the same structure and complete t...
Need someone expert in Linux to build us an iso that will install freepbx in Ubuntu 22.04 with latest php 8.1 and python3.
...language and dialect. Example: Portuguese (Brazil) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the Portuguese language, the file uses English words, then you need to put "Portuguese (Portugal), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your n...
I am looking for a developer who can integrate Whatsapp messaging with freepbx UCP using a third party API. The project requires the following features: - Ability to send and receive text/media files through Whatsapp - The system does not need to handle multiple languages - The budget for this project is between $100-$500 Ideal skills and experience for this job include: - Strong knowledge of Whatsapp API and freepbx UCP - Experience in integrating third party APIs like twilio/ - Proficiency in programming languages such as PHP or Python If you have the necessary skills and experience, please submit your proposal with relevant examples of your previous work.
Implement real-time phone speech recognition in project with Asterisk PBX and Kaldi/Vosk experience with Asterisk PBX and Kaldi/Vosk
...project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, ...
...project with Asterisk PBX and Kaldi/Vosk. I use Asterisk-specific module () to carry out ASR operations without compatibility issues. So far if anybody speaks anything while calling, it gives very clear text output. The problem I'm struggling is how to enable streaming ASR immediately during the conversation, i.e. since Dial() application of Asterisk dialplan gets executed. That's a subject of this job - create script (most likely, with some Asterisk REST Interface components) which works as follows: 1) since Dial() application starts running, real-time audio stream gets processed via ASR engine that is waiting for inputs inside of docker container (because I deploy Kaldi as a software built in Vosk server which is compatible with Asterisk, ...
We need our asterisk to play a small audio file to destination at the moment we receive 180 or 183 from destination. Destination provider needs 5 RTP packets to discover IP and port when asterisk is behind NAT.
...WhatsApp number. - We will provide the phone number/phone numbers and pictures for the WhatsApp account. - The project should be multi-channel. I would like to able to start multi-calls ( you can run multi WhatsApp account with multi-phone number or can use just one WhatsApp account with multi calls. ) - The development platform/operating system is not important. you can use Asterix, Freeswitch, FreePBX vs... - The implementation should return the correct call error codes to the SIP backend like CALL SUCCESS(200 OK), BUSY(486 Busy Here), UNAVAILABLE(503 Service Unavailable), etc.... to try other rounds on the other SIP Switch. Functional flow 1) Calls from PBX/sip gateway will be sent to WhatsApp gateway 2) WhatsApp gateway converts the SIP to WhatsApp protocol 3) If it i...
We want to create a custom cdr module that we can control the extensions that will appear in the report. For example, we want to have the ability to show/hide specific extension from appearing in the cdr report as if we want to allow specific extensions to appear in the report according to the privileges of the logged in user from the freepbx web admin portal. So, we can assign one or more extension(/s) to a specific user and this user will only be able to search and find calls details and listen to the recording of these assigned extension(/s) to his/her account and can't see any other call details for any other extension(/s) that is/are not assigned to the account. Also, we want to allow only listen and download of the recording without having the ability to delete it.
I am looking for a skilled freelancer who can help me change the operating system of my Openstage 20E Siemens IP Phone to SIP in order to work with Asterisk PBX. Key Requirements: - Experience in configuring IP phones and working with Asterisk PBX - Proficiency in uploading SIP firmware to the IP phone - Familiarity with Openstage 20E Siemens IP Phone and its operating system Specific Features: - Enable voicemail, call forwarding, and automated attendant on the SIP phone Preferred Method of Communication: - Remote access to the phone for configuration - Step-by-step instructions via email for guidance - Phone call or video conference for any necessary guidance If you have the necessary skills and experience to complete this project, please submit your proposal.
We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/ all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.
I am looking for a freelancer to help me make outbound calls using my existing FreePBX system. The calls will be related to customer support and I need help setting up to make out bound calls that i can change the caller id for. On average, I expect to make 50-100 outbound calls per day.
Final Step for Running Asterisk and Freepbx with door opening script.
OpenStage 20E SIP Firmware I am looking for a skilled professional who can assist me with installing the latest version of OpenStage 20E SIP Firmware. Requirements: - Experience with OpenStage 20E SIP Firmware installation - Familiarity with Asterisk PJSIP Specifics: - The current version of the firmware is not specified, so the freelancer should be able to handle any version (1.0, 2.0, or 3.0) - The main goal is to ensure compatibility with Asterisk PJSIP - The project requires immediate attention, so the freelancer should be available to start working on it right away Deliverables: - Successful installation of the new firmware - Verification of improved audio quality, enhanced security features, and bug fixes/stability improvements If you have the necessary skills and...
this quote if for setup and configuration of an Asterisk Server with EC20 module.
I am looking for a freelancer who can assist me in loading Asterisk on my Linux server. The ideal candidate for this project should have the following skills and experience: - Proficiency in Linux operating system - Experience with setting up and configuring Asterisk on Linux servers - Familiarity with cloud servers, specifically working with Asterisk on a cloud server environment - Knowledge of SSH/Telnet connection for remote access and configuration The main objectives of this project include: - Installing and configuring Asterisk on the Linux cloud server - Ensuring proper integration and functionality of Asterisk with the server - Setting up SSH/Telnet connection for remote access and configuration If you have the necessary skills and experience,...
...language and dialect. Example: Bulgarian (Balcan) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the Bulgarian language, the file uses English words, then you need to put "Bulgarian (Rup), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your notes, wh...
We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/ all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.
I would add another office to our exsting freepbx server having two different locations.
I need some tech help to add another office to our exsting freepbx server. Two different locations.
...can help me with creating an Asterisk prototype in Ubuntu Docker. The project requires the following functionality and features: Specific functionality: - A voicemail system that allows users to leave messages. It has to run inside an ubuntu docker. Additional requirements: - Integration of Fail2Ban to enhance security and protect against unauthorized access - Implementation of PJSIP for improved audio quality and performance - Do not use FreePBX or any other Bloatware. Your solution will be rejected and you will not be paid if you do that. Has to be done only via asterisk. - Demonstrate working inbound voicemail service that plays a greeting and accepts voicemail on Telyx SIP lines. Ideal skills and experience: - Strong knowledge and experience with Asteris...
...expertise of the freelancer to determine the best configuration. Regarding the authentication method for the SIP server/client, I am not sure if a secure authentication method is required. Further clarification on this matter would be appreciated. Ideal Skills and Experience: - Experience with configuring Zenitel tcis3 system - Familiarity with Windows-based SIP server/client software, such as Asterisk, FreeSWITCH, or 3CX - Strong knowledge of SIP protocols and configurations - Ability to provide recommendations and suggestions based on project requirements The systems will be for a internal/private network. So should not be reliable on a internet connection. So no cloud solutions. This is a basic setup so tcis3 can call windows pc and Windows pc can call tcis3. Nothing el...
We are looking for Click to conference Solutions.. I need support for "Click to conference Solutions" we would like to Dinstar GSM Voip Gatway. Step 1 I will call API then it should make a conference call. Eg: https://astersik-ip/ all callers should connect to conference call. Step 2 Should get announcement in the call eg: caller1 connected, caller 2 connected, caller 3 connected ..... .... Step 3 When people on the conference call, if we need to add another person. We should be able to add by tying *333 on the Dialpad DTMF keys. in the database we will pre-defined varibles. 333 --> 9848424242 444 --> 9858585858 Step 4 Call Recording.
Looking for a sip soft phone that can connect to asterisk and make calls. It will use codec g729. Call history, settings and other standard features.
Hi, We have setup a video conferencing system using Asterisk and using ConfBridge application of Asterisk to manage the calls. In that, we are facing audio echo issue in both audio and video calls. Need help to get this resolved asap. PLEASE BID ONLY IF YOU HAVE GOOD EXPERIENCE WITH ASTERISK AUDIO AND VIDEO CALLS USING WEBRTC.
I need software to be burned in my chipset, the devloper should have chinese embeded system, Should have prior knowledge of configuring and setting ASTRIX PBX SYSTEM Need a software to be devloped, project based on asterisk for embedded system before.. Major knowledge in VOIP Prefered operating system should be LINUX
...rodsrch **The steps below outline the process to create a variation family via 1x1 upload:** 1. Select 'Add a Product' from 'Inventory' drop-down. 2. Click on 'I’m adding a product not sold on Amazon'. 3. Search or browse for the category that matches the product you want to sell and click the 'Select category' button. 4. On the 'Vital info' tab, complete the required fields (marked with an asterisk *), and click on 'Continue'. 5. In the next tab of 'Variations', select a theme from the 'Variation Theme' drop-down list. 6. Select the appropriate variation theme that you think is relevant for the products you are listing. Note: The Variation Theme drop-down list and Variations tab will appear o...
I need help with adding reporting from a CSV file to Freepbx. I'm not sure if I need assistance with retrieving the data and I do need to integrate the data with an existing Freepbx module. My plan is to generate reports from the saved CSV file. The project will require considerable knowledge of Freepbx and the ability to integrate the data into an existing Freepbx module. The project should also include methods for data validation and analysis. If needed, I may be open to additional advice and guidance regarding how to manage this data effectively.
We are looking for a Chinese embedded system developer to make Asterisk PBX software Preferred Operating System: Linux Project Completion Time: No time limit Ideal Skills and Experience: - Proficiency in Linux operating system - Experience in setting up and configuring Asterisk PBX system - Knowledge of auto attendant, call routing, recording, and voicemail setups - Familiarity with email forwarding configurations PBX software must be able to be burned on an embedded system
We are an existing business in need of an Asterisk setup for PBX CRM integration. We already have a CRM system in place and are looking for someone with experience in integrating it with an Asterisk PBX. The ideal candidate should have the following skills and experience: - Experience in setting up and configuring Asterisk PBX - Experience in integrating Asterisk PBX with CRM systems - Familiarity with popular CRM systems such as Salesforce, Zoho, and SugarCRM - Strong communication skills and the ability to work collaboratively with our team As there is no time limit, the project can be completed at a reasonable pace.
We are an existing business in need of an Asterisk setup for PBX CRM integration. We already have a CRM system in place and are looking for someone with experience in integrating it with an Asterisk PBX. The ideal candidate should have the following skills and experience: - Experience in setting up and configuring Asterisk PBX - Experience in integrating Asterisk PBX with CRM systems - Familiarity with popular CRM systems such as Salesforce, Zoho, and SugarCRM - Strong communication skills and the ability to work collaboratively with our team As there is no time limit, the project can be completed at a reasonable pace.
Project Description: features to implement in Asterisk to handle call traffic more efficiently to avoid getting GSM cards blocked, (Issabel installed and Skyline 32/132 GSM Gateway available, and current on use platform where VOIP to GSM working perfectly ) Skills Required: Incorporates all the mentioned features Call Throttling Time-of-Day Routing Gosub and Subroutine Traffic Shaping and Qos Load Balancing Call Queues Limit Concurrent Calls Custom Dialplan Logic Project Requirements: - Assistance is needed in setting up a custom dial plan Ideal Freelancer: - Has experience in working with Issabel, Asterisk, and Skyline GSM Gateway - Familiarity with VoIP and GSM Gateways, and their configuration and management Note: Please include your relevant experience and expertis...
I am looking for a solution for our Voip webphone telephony. We have our own Asterisk server and receive calls via our CRM and can also make calls over it. Now we want to add more features: - It shows history, who called and when. How long the phone call went on - Click to call functions. In our CRM we want to click on the number and it should connect directly to it - We would also like to see when someone calls, if it is displayed in our database, i.e. CRM Ideally, you may have call center functionality to build in. We serve 3 numbers at the moment. For CRM we use a bootstrapframework php 7.4 Can you make our offer for it.
...language and dialect. Example: Chinese (Standard) c) if, in addition to the main language, 1-2 words from another language are used in the file (not full-fledged phrases, but just words), then you need to specify the name of the second language with an asterisk. If the second language is not familiar to you, then you need to put "unknown language". Example: in addition to the Chinese language, the file uses English words, then you need to put "Chinese (Min), English*"; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be indicated. In the first place it is necessary to indicate the language prevailing in the audio. In the "Comments" field, you can specify your notes, which ...
I am looking for a solution for our Voip Webinterface telephony. We have our own Asterisk server and receive calls via our CRM and can also make calls over it. Now we want to add more features: - It shows history, who called and when. How long the phone call went on - Click to call functions. In our CRM we want to click on the number and it should connect directly to it - We would also like to see when someone calls, if it is displayed in our database, i.e. CRM Ideally, you may have call center functionality to build in. We serve 3 numbers at the moment. For CRM we use a bootstrapframework php 7.4 Can you make our offer for it.
