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    461 dialplan tugasan ditemui, harga dalam USD

    ...Compatibility: Implement gRPC to ensure compatibility for seamless communication between components. 3)Real-time Transcoding: Enable real-time transcoding capabilities to convert audio data appropriately for interaction with OpenAI Whisper. 4)Asterisk Integration: Integrate the solution with Asterisk to allow seamless initiation and handling of audio calls from Asterisk dialplans. Example Asterisk Dialplan: [audiosocket] exten = s,1,Answer() same = n,AGI() same = n,MixMonitor(/var/www/html/calltest/${uuid}.wav) same = n,Ringing(3) same = n,Wait(1) ;same = n,Playback(silence/1) ;same = n,Playback(custom/audioStream) same = n,Dial(AudioSocket/${uuid}) same = n,Verbose(0,Result was ${SPEECH_TEXT(0)}) same = n,Hangup() exten = h,1,Hangup() Deliverables: 1)Fully functional

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    I'm currently using a private VoIP provider for my Op...Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation, ...

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    I'm currently using a private VoIP provider for my Op...Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation, ...

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    Hi there, I'm in need of assistance to modify my Vicidial dialplan. The key changes I need are: - Call Routing Changes: I need to implement a system that prioritizes certain calls. However, the basis for this prioritization hasn't been determined yet and your expert opinion is highly valued. Key skills and experiences: - Extensive experience with Vicidial dialplan modifications. - Understanding of different call routing strategies. - Ability to suggest optimal call prioritization methods based on business needs. - Proficient with creating custom IVR prompts. - Familiarity with modifying the agent’s screen for better user experience. Your task will be to provide a solution to prioritize calls and implement it within the existing system. Your advice is soug...

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    I'm in need of an Asterisk expert to configure a dialplan for my business. The core functionality required includes: - Attended Transfer & Whisper Transfer Dialplan: This specialized functionality should be available for agents / extensions . Skills and experience preferred: - Expert in Asterisk PBX configuration - Proven experience in setting up dialplans with the above-mentioned features. Your proposal should demonstrate your ability to carry out this project successfully. Note that the ability to meet timelines and provide post-delivery support will be a significant advantage.

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    mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

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    Hi I'm looking for an Asterisk AGI written in GO that is probably going to use this library: and which is called from the dialplan as: exten => 500,1,AGI(gotest,${myVar}) exten => 500,n,HangUp and is able to: * read the 'myVar' variable * read the 'agi_extension' * print to syslog and exit if some variables are missing * execute a saydigit(123) * execute the playback of a wav file * use get_data to get a digit and log it to syslog * set the callerid to 456 * execute a dial(SIP/789) with max ringing 60 seconds and return the ANSWEREDTIME and DIALSTATUS arrays * hangup max bid is 100 euros you must have your own Asterisk setup and GO environment and provide instructions on how to setup and build the code.

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    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 handle high call volumes efficiently, ensuring a smooth and effective sales proc...

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    ...in 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, here is

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    ...in 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, here is

    $104 (Avg Bid)
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    Project Description: features to implement in Asteris...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 expertise in your...

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    ...needs some integration into Asterisk software, 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, here is

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    11 bida

    ...needs some integration into Asterisk software, 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, here is

    $265 (Avg Bid)
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    10 bida

    Project Description: Asterisk PBX DialPlan Customization I am looking for someone with experience in Asterisk DialPlan customization to help me customize my existing DialPlan. The ideal candidate should have knowledge and experience in Asterisk DialPlan scripting, as well as the ability to understand and modify existing DialPlan code. Skills and Experience: - Strong understanding of Asterisk DialPlan scripting - Experience in customizing existing DialPlans - Ability to troubleshoot and debug existing DialPlans - Familiarity with Asterisk configuration and setup - Knowledge of VoIP technologies and protocols This project does not require assistance with other tasks besides the DialPlan customization, and there is no need for assistance wit...

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    Project Details: We are looking for someone who can assist us with forwarding the audio from the Asterisk dialplan to a server in the backend for live audio transcription of every call. Our team will handle the configuration of the Asterisk server itself, so we mainly need your consulting and "know-how" skills to ensure a smooth connection and provide guidance. Additionally, having a sample server showcasing the connection would be beneficial. If you have experience working with Asterisk and possess the knowledge to facilitate this audio forwarding and transcription setup, I would be very interested in discussing the project further with you. Alternatively, if you know someone in your network who has the necessary skills and might be interested in this opportunity. Pleas...

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    ...a HTTP GET will be used to retrieve the contents of the provided url. The value written to the function specifies the destination file of the cURL'd resource. Example: Retrieving a file 1 exten => s,1,Set(CURL(http://localhost:8088/static/)=/var/spool/asterisk/tmp/)) Note Icon If live_dangerously in is set to no, this function can only be written to from the dialplan, and not directly from external protocols. Read operations are unaffected. Syntax CURL(url,post-data) Arguments 1. url - The full URL for the resource to retrieve. 2. post-data - Read Only If specified, an HTTP POST will be performed with the content of post-data, instead of an HTTP GET (default). See Also 1. Asterisk 16 Function_CURLOPT ...

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    Hi, I have goautodial 4 setup on server with ssl/tls everything is working great except for the carrier dial plan that does not make calls out. All i need is assistance with the dialplan to make sure that the calls can be made from the webrtc phone built into goautodial 4. The software is setup already the solutions looks like it works fine. I assume the problem is with the dial plan. I need someone that can go through the system to make sure calls can be made out using the web phone built in.

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    We need an Asterisk 11 dialplan context that: 1. Executes when dialing any number from the extension 1005 and also from extensions 1070 through 1078 and 1081 through 1089 2. Blocks the call and plays the "number-not-in-service" playback file if the number dialed by the extensions above is equal to the result of the query "SELECT mobile FROM vtiger_contactdetails WHERE firstname REGEXP '^[0-9]+$' and mobile = ${EXTEN: -8}". The query is done to a localhost MySQL database

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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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    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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    Ditampilkan
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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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    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...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 di...

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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 fiel...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 amnd last three data pulled out to select to go into the dispatch panel Deploy FreePBX 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 di...

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    ...Asterisk, Freeswitch, Opensips, Kamailio VOIP, SIP, IMS, NGN, ISDN, TDM, and Telecom/Network Protocols. · Very Good Knowledge of VOIP/SIP servers, Design and Development, Support, Testing, Deployment, Asterisk Programming, Asterisk Administration, FreeSWITCH Dialplan, Freeswitch Administration, LUA programming, Opensips/Kamailio script Programming and good exposure to VoIP Gateways / Servers / Computer Telephony Integration, using signaling protocols, SIP, and media protocols, IVR Programming, and Customization in the Fusionpbx in the Dialplan , Core modules and AGI/AMI programming using C/Python /Perl/Shell. · Working experience with products like FreePBX, A2billing, FusionPBX, OPenIMScore, and Vicidialer · Design and Development of VOIP solutions, VO...

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    mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

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

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

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    Someone must get experience in FreePBX and Asterisk. Main task is making dialplan , and solve some small issues. if someone have rich experience in this field, it will not take 3 - 5 hours. Long Term Project. As based on this result, we can work continually.

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    mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

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    ...would have an access to the billing portal through and they format of displaying client billing information is subject to the requirements 3) PBX portal - This is where the real work happens, building the pbx requires your knowledge in running native base code such as Golang to create api with connecting feature to the pbx system. The pbx module would be the pbx system dialplan design using asterisk and it should generate dialplan base on the asterisk code when that front end on the self service request configuration from the pbx. 4) Payment gateway - The portal must facilitate with payment gateway and it allows clients to see payment history on a click away. The historical data consist of their subscription and call usage where they can refer to instantly and this payme...

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    need to make dialplan for realtime queue using mysql for asterisk system

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    mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

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    OpenSIPs Admin Tamat left

    mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

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    This requires setting up a call center with FusionPBX/FreeSwitch with the following points considered - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

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    I need someone to configure a Freeswitch/FusionPBX server. It needs: - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

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    Segera
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    Freepbx expert Tamat left

    we need an expert in freepbx to help us with dialplan

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    Hello I have a astpp open source VoIP server. ASTPP using LUA script as a dialplan. I need to fix one bug in that script. So looking for someone who is very expert in LUA. If you even worked with ASTPP lua scripts then it will more better then you will understand what I want to fix. Note to Freelancer.com: I have post yesterday the similar project. Just can't understand why it was rejected. Astpp is open source and anyone can modify by their own. Please see Please approve the post and you can delete those admin notes

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    Asterisk development, creating dialplan and ivr desgin Vicidial Intergration

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    This is a new Asterisk 18 installation from scratch, preferred a Docker/Podman container. For this project, you need to set up a new IAX trunk with one of our partners and set up the dialplan as per the specifications received. Basic calls routing from incoming to less than 10 VoIP SIP phones.

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    This project contains 4 milestones: Milestone 1 Setup fully working Asterisk 18 as Docker container with ODBC MySQL remote connection: Dockerfile Asterisk config fil...Asterisk 11.7.0 to Asterisk 18 container from Milestone 1 with MySQL connection to a remote MySQL server/container. Milestone 3 Review existing design/architecture of PHP applications - Asterisk interaction and provide feedback, improvements and best practices. Milestone 4 Install and configure a new SIP connection with a remote 4 SIM cards GSM gateway and configure the dialplan to add this SIP connection into the existing dialplan and application: set priorities between existing SIP trunks, set failover (to use the next SIP trunk if the first one is not available etc) If more information or details are n...

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    ...voice communication. This is an Asterisk module that implements an asterisk application that can be used from dialplan. It should be able to change human voice of a speakers in a quality way. It should be able to manipulate voice attributes like: chiefly frequency, harmonic structure, intensity... It also must be capable to identify and change formant of human voice. Resultant real-time voice must have the following characteristics: it is clear, it keeps intimacy of a speaker and it is natural. DSP should be able to produce at least 10 different resultant voices from a single speaker. It should be possible to set all parameters that define voice attributes when called from a dialplan. Every channels should have its own DSP component. There is an existing Voice Changer aste...

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    Looking for someone that knows asterisk and knows how to code in C. Looking to fix chan_alsa to support sending calls to more then a single alsa device including virtual devices from within the dialplan. The idea is the virtual devices will be setup to stream to AES67 receivers and allow overhead paging. Anyone that also knows AES67 is a major plus

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    ...una integración webphone de asterisk. Con la cual se pueda realizar llamadas de un área a otra, de una persona a otra, de un anexo al exterior. Entre las funcionalidades que debería contar Asterisk es: * Grabación de llamadas * Límitador de llamadas * Ocultamiento de número celular destino * Habilitación de soporte web para el servicio Asterisk * Habilitación de soporte webrtc * Programación de dialplan para click2call * Configuración de Troncal SIP con operador * Programación de soporte para anexos dinámicos y temporales * Creación de servicio web para generación de anexos temporales y dinámicos * Creación de script de consumo de servicio web para devolución de...

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    I need an Asterisk Expert for couple of works. More details will be described later. You need to be able to create asterisk scripts and have to be expert on making dialplan.

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    I am looking for a OpenSIPS expert to update my opensips from Version 2.2.7 to Latest stable release Version 3.1.1. You need to make sure everything is working. Including the dialplan and database.

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    ...needs some integration into Asterisk software, 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, here is

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    1 bida

    Major info in a meeting call once project is given, Consulance and creation custom DialPlan

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