Kannel asterisk jobber

I am looking for experienced devOps that can work on setting up Asterisk, implementing Vosk STT, and setting up TTS. I have a server in place for the project, so devOps with expertise in this required infrastructure is of utmost importance. The ideal candidate should have good skills in Asterisk, Vosk STT, and TTS in order to execute the project successfully and to my satisfaction. Only applications from expert level devOps will be accepted.

...this language and dialect. Example: French (Quebec) 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 French language, the file uses English words, then you need to put "French (Paris), 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, whic...

Need to install and then configure so I can receive calls. Immediate work.

...language and dialect. Example: Japanese (Hachijō) 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 Japanese language, the file uses English words, then you need to put "Japanese (Kyūshū), 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, w...

**Project Description:** **Overview:** We are in need of an experienced system administrator with expertise in VoIP, Asterisk PBX, and Linux to configure our GoIP8 device. Our goal is to set up GoIP8 as a SIP trunk within FreePBX 14. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in configuring GoIP devices and SIP trunks. - Examples of previous projects where you successfully configured similar VoIP setups. - An outline of your approach to configuring GoIP8 as a SIP trunk within FreePBX 14. - Your proposed timeline for completing the configuration. - Your pricing structure for this configuration project. **Note:** We are looking for a reliable and efficient configuration of GoIP8 as...

I have Job Listings, from Indeed, in Workbook, on SHEET 1. SHEET 2, i want to have/use, 3 Columns for FILTER the data in SHEET 1. COL_1=Job Title or Position, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER (add/approve/allow) from SHEET 1 to SHEET 3, final OUTPUT SHEET COL_2=Company or Business Name, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER OUT (remove/ignore/delete) from SHEET 1, to SHEET 3, final output SHEET. COL_3=Location, I want to ADD key words (even use of ASTERISK) in this COLUMN to FILTER (add/approve/allow) from SHEET 1 to SHEET 3, final OUTPUT SHEET SHEET 3 - is the OUTPUT, from ALL DATA in SHEET 1, using the SHEET 2 Filters (3 columns with a FILTER/APPLY button), and SHEET 3 ends up with the res...

Necesito configurar un FREEPBX para tomar una trama SIP de 30 canales y 10 de estos canales pasarlos a otro FREBPX a una cola de llamadas, otras 3 líneas pasarlas a internos de este segundo FREEPBX. El resto de las líneas se conectaran a un tercer FREEPBX.

**Project Description:** **Overview:** We are seeking a skilled developer to integrate Asterisk PBX into our system for the purpose of answering incoming calls, transcribing audio to text, and converting text to speech. It is essential that these processes are performed offline, without reliance on external hosted solutions like Google. **Submission Requirements:** Please provide the following in your proposal: - A summary of your relevant experience and expertise in Asterisk, C programming, and offline audio processing. - Examples of previous projects or work that demonstrate your skills in these areas. - An overview of your approach to achieving the specified objectives. - Your proposed timeline for completing the integration. - Your pricing structure for t...

I am looking for a freelancer who can troubleshoot my Issabel (Asterisk) configuration with GoIP. Specifically, I am experiencing connection issues and there are error messages that need to be addressed. Ideal skills and experience for this job include: - Proficiency in Issabel (Asterisk) configuration - Experience with GoIP configuration - Knowledge of troubleshooting connection issues - Ability to address and resolve error messages. The successful candidate should solve any configuration issue with Goip to handle in-out connections

I'm looking for an experienced software development team to help build a custom Asterisk or Freeswitch speech-to-text system for my company. This system needs to have the ability to convert live speech and respond back by pressing DTMF digits. (Example: inbound call will play "press 3 to continue" ,at which point your software would press the digit 3 or whatever digit is in the initial spoken phrase. Then, the inbound call (assuming you pressed the right digit) will say the next phrase. Whatever is said in that next phrase would have to be speech2text converted and saved to db & HTTP POSTED to a remote url). I am looking for this project to use cepstral or some other free speech2text software - not looking for paid APIs) If you have experience with similar projec...

I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Kamailio - Configure TLS certificates, will need to work with wildcard cert - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk

I am looking for a freelancer who can help me integrate video-calling functi...video-call is to enable communication between two extensions on a Raspberry Pi using Node-red code. I already have a script with audio-calling, but i need to make it video-calling with VP8 codec. I need someone that know what he/she is doing...I can provide the script of audio-calling, to help him/her change it. I will pay 50 dollar. Skills and experience needed: - Proficiency in Node-red and Asterisk - Knowledge of video-calling protocols and codecs - Experience in integrating video calling functionality into existing systems Preferred libraries and tools: - The client is open to suggestions for libraries and tools to use in this project. Timeline: - The client expects the project to be completed with...

1st phase Currently, the active server is in 131 . Install and configure the new version on 130. After all done, change the IP from 130 to 131 (Change the IP from 131 to 130). And wait 1-2 days to see if any clients have complaints or no problem:). ( at least we have backup 131 old sever we make it 130 for backup) 2nd phase After 1-2 days good to go then we do upgrade new version and config to 130 (previously 131) After all is done, change the IP from 130 to 131 (Change the IP from 131 to 130 again). And wait 1-2 days to see if any clients have complained. 3rd phase Then we can do a load balance between 131 and 130. ( to test load balance ) 4th phase Step by Step installation and configuration video from 1st phase to 3rd phase

Build a web GUI for Asterisk 20 from scratch Requirements: - Programming language: C, Python, or any language that can fit the project. - Specific features: understanding asterisk - Desired timeline: Within 3 months We are looking to build a web GUI for Asterisk 20 from scratch. The GUI should have all asterisk features. We develop asterisk codes and config files by ourselves and need to build web GUI to allow users to manage asterisk features - The project should be completed within 3 months. Ideal Skills and Experience: - Strong proficiency in C, Python, or any language that can fit the project. - Experience with Asterisk and building web GUIs for telephony systems. - Ability to work within a specified timeline and deli...

...Registered users will have the option to fill in the following information on a dedicated website: Domain Page URL for displaying the message (default: entry page of the site, the first opened page) (marked with an asterisk, meaning all pages of the domain, zero implies none) The message will contain the words "Payment with DCP Cash is available on this site" (modifiable) and any additional information site owners may want to add, such as a 10% discount for DCP payments. Wallet owner's name Wallet identification The address where the payment window will be displayed (marked with an asterisk, meaning all pages of the domain) Payment message text LocalStorage path for the payment amount LocalStorage path for currency (USD, DCP, or STAS) The address to which t...

I am seeking an experienced developer to assist with the deployment debugging of an Asterisk based CRM. The specific issue that needs to be resolved is integration problems with other systems. Skills and Experience Required: - Proficiency in Asterisk and CRM systems - Strong knowledge of integrating systems with databases, third-party software, and hardware - Experience in troubleshooting and resolving integration issues - Ability to work under pressure and meet tight deadlines, as this project is of high priority.

I am looking for a developer to develop a call forwarding pannel and pbx panel on freeswitch voip. The call forwarding panel and pbx panel should include advanced call forwarding with scheduling and should include integrated voicemail and call recording. I am not sure which VoIP provider to use and would appreciate advice in this regard. I ...appreciate advice in this regard. I am targeting completion within a month. The successful developer will be knowledgeable and experienced in using freeswitch VoIP in order to complete the project. If this project is completed to my satisfaction, I will consider having the developer complete additional projects with me. Looking forward to hearing from an interested and qualified candidate. No asterisk Candidate. Only Freeswitch other voip devel...

I need asterisk to do the following: 1. asterisk Dial to callee 2. Callee pick up the call and the call is answered/ bridged 3. After X seconds, asterisk will inject DTMF tone 4. Caller will not be able to hear the DTMF.

I am looking for a freelancer who can help me with installing and reconfiguring Kannel and the database Postgres SQL for SMS messaging. Current Status: - The Kannel installation is already done but it is only partially configured. Specific Configurations: - I have a general idea of the configurations I need for my Kannel installation. SMS Volume: - I anticipate sending more than 500 SMS messages per day. Ideal Skills and Experience: - Experience with Kannel installation and configuration - Knowledge of Postgres SQL and database management - Familiarity with SMS messaging systems - Strong troubleshooting and problem-solving skills

Hi Aqs Y., I'm looking for a consultant to upgrade from freeswitch 1.10.6 to the latest on debian 10 (most probably also debian needs to be upgraded, to 11 or 12, as you prefer) It needs to work with asterisk 16.29.0 and also needs to support mod_expr, ceil and randomize

We are seeking an experienced freelancer to facilitate the seamless integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and ...integration of WebRTC-based calling capabilities into our Isaaabel SIP account. Isaaabel currently operates with UDP as its SIP protocol, and our primary focus is to ensure the utmost security for our call traffic through Encrypted Media. Project Requirements: -Integration of WebRTC functionality into the Isaaabel platform. -Configuration of JsSIP npm package to establish connections with our Asterisk calls to SIP and PJSIP extensions through the WebRTC interface. -Implementation of robust security measures, including encrypted media, to safeguard call communications.

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I am looking for a freelancer who can help fix the issue with the Vicidial inbound carrier stats for my existing business. The specific problem we are experiencing is missing data in the reporting. We need this issue to be resolved within one day. Skills and experience required for this project: - Expertise in Vicidial and inbound carrier stats - Strong troubleshooting and problem-solving skills - Experience with data analysis and reporting - Familiarity with call center operations - Ability to work efficiently and meet tight deadlines

I am in need of assistance with a clean up and addition of features to an Asterisk PBX system. Specifically, I'd like assistance with setting up an IVR menu which can be used to direct customer calls to the right departments. Additionally, I would need the system to be maintained regularly after installation of the features. I have a preferred method of communication through email and am eager to find an experienced freelancer for the task.

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I am looking for a freelancer to help me integrate a WebRTC based calling experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encry...experience with my Isaaabel's SIP account. Isaaabel supports UDP as the protocol for SIP account, and I need the experienced freelancer to ensure security of the calls with Encrypted Media. This is an urgent requirement so it would be great if the freelancer can deliver the project quickly. Resume, we need to connect WebRTC extension with Issabel. I need that JsSIP npm package can connect with my asterisk server and make calls to SIP and PJSIP extensions. I need the documentation of the implementation for future re-ins...

I am looking for an experienced Asterisk developer to create a web interface to manage all of its features. Specifically, I need call routing and forwarding, an Interactive Voice Response (IVR) system, and call recording and monitoring capabilities. No specific requirements for the web interface are necessary and the freelancer will not be expected to provide post-development maintenance or support.

...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 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. Specif...PBX 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 projec...

...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...

...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 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...

...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...

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.

Se necesita tecnico experto para montar remotamente una central asterisk, que pueda hacer llamadas locales usando un modem 3g con chip celular. Configurar softphones para personal.

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.

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...

I am looking for a freelancer who can assist me with installing and reconfiguring kannel and the PostgreSQL database for my SMS system. Current Setup: - I already have both kannel and PostgreSQL installed on my system. Additional Hardware: - No additional hardware is needed for this project. Specific Features Needed: - I require an SMS gateway for my SMS system. Ideal Skills and Experience: - Experience with installing and configuring kannel and PostgreSQL. - Knowledge of SMS gateways and their setup. - Understanding of database management, specifically with PostgreSQL. - Ability to troubleshoot and resolve any issues that may arise during installation and configuration.

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.