hi i have run voip example on esp32 lyraT kit and used local sip server(minisip) then it is working fine for call , but i have hosted the asterisk sip server on goolge cloud (the asterisk is working fine as i tested by calling using mobile apps. ) but when esp32 connects with this asterisk server whenever i call from mobile app upon pressing play button it says " no body is available to attend your call" .
I am looking for a freelancer to help me with the installation and configuration of a FusionPBX cluster on a Linux ...me with the installation and configuration of a FusionPBX cluster on a Linux server environment. I do not require data migration as part of the installation. However, I am unsure if I will need ongoing maintenance and support after the installation. Ideal Skills and Experience: - Extensive experience in installing and configuring FusionPBX clusters on Linux servers - Strong knowledge of VoIP and PBX systems - Familiarity with Linux server environments and command line interfaces - Understanding of network architecture and security protocols - Ability to troubleshoot and resolve technical issues related to FusionPBX clusters - Excellent communication and problem-solv...
I have a server with an FXO card, the scenario that i need to do it is this: 1.- i have extension 100 with a mobile phone in a follow me destination 2.- if the mobile phone it is not picked up in 9 or 10 seconds needs to send to voicemail or IVR or another destination (like...scenario that i need to do it is this: 1.- i have extension 100 with a mobile phone in a follow me destination 2.- if the mobile phone it is not picked up in 9 or 10 seconds needs to send to voicemail or IVR or another destination (like a ring group) the main problem is that i cannot make the asterisk to understand when the PSTN call exceeds that ring time because for asterisk the call it is answered so i need a help to setup correctly my box for make it to work please only experienced freepbx / a...
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.
...======================================================= if you did a similar project before please share with us some pictures and information. you will be our priority. Hi, The WhatsApp channel allows consumers to connect with brands and brands to connect with consumers from the WhatsApp app on their smartphone. We need to develop a SIP to Whatsapp voice gateway to send some of the calls from our PBX. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the destination's 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-...
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.
We want to have a tool that can deploy easily new instances of PBX. In each instance we will need to have a panel where admin of instance will be able to control following: - add/ remove extensions - lanch calls and choose SIP lines to make the call - active/ quit recording - set Ivr - set queue - Add/ remove SIP lines We will need to control the panel by Api and be able to send calls by api with parameter instance, extension and phone number to reach and it should call the extension to start the call. we will also need an endpoint to active/ deactivate instance and if Instance is off, user could not access his portal.
Configure a GrandStream PBX model UCM 6204 with 6 extensions
...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 G...
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 structur...
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...
I'm in need of an experienced service provider to install and configure a PBX system with 25 extensions. The system I need is an on-premise one which does not require integration with CRM or any other software. I expect the service provider to be well versed in PBX installation and configuration and have experience with similar systems. The project requires all necessary supplies, installation, and configuration of the PBX system with 25 extensions. Upon completion of the job, I would like to be fully operational with the new system.
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
We would like to install a fresh FusionPBX on VPS server what we need it secure: a. Firewall b. SSL / TLS (Let’s Encrypt) c. XML RPC d. Fail2ban e. Sip registration from nated networks issues like sorting out one way voice etc. Do test make sure everything works for multitenent clients etc. . Please message me for more inquiry that i need. Thank you.
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 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 developer only.
I have a VOIP PBX based on Freepbx with 2 extension . 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 activated with the night and day option with code *280
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.
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.
I am seeking an expert in Plivo VOIP and PBX IVR configuration to assist me with my project. Specifically, I need assistance with setting up IVR menus and configuring call routing. For call routing, I do not have a preferred strategy, so I am open to suggestions. I need someone to setup to do IVR, setup greeting message, music on hold, voicemail, hours of operation routing and etc. If you are hired, I will provide Diagrams and prerecorded audio files. You must have all necessary skills and experience working with Plivo. Calls will be routed to a mobile app that you recommend unless Plivo has one. Ideal skills and experience for this job include: - Extensive knowledge of Plivo VOIP and PBX IVR configuration - Experience in setting up IVR menus - Proficiency in confi...
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.
I have been using a voip system that was setup by someone else. Recently it stopped working. I tried to login to fusion PBX but can't login anymore it says password not recognised. All other features of the system are working i.e domain name, telynx all ok need someone to help get it restarted
I am currently facing an issue with my Issabel PBX setup connected to a D-Link DVG-6004S FXO Gateway. While I'm able to make outgoing calls successfully, I cannot receive incoming calls. Requirements: 1- Proficiency with Issabel PBX and D-Link FXO Gateways. 2- Ability to quickly diagnose and resolve the issue. 3- Availability to work on this immediately and resolve within 50 minutes. 4- Strong communication skills in English. If you believe you're the right fit, please contact me ASAP. Compensation will be provided for the quick and successful resolution of this issue.
Please don't bid if your bid request is more than 20 usd. We are looking for someone for installing Ubuntu Linux 22 and free pbx. Also configure apache for multiple domains and subdomain with latest php 8.1+. At the end provide a iso image that can be used to reinstall in any x64 machine.
I am looking for a skilled developer to create a Cloud-based PBX system with the following requirements: Features: - Call Recording capability - Auto Attendant for efficient call routing - Voicemail to Email functionality for convenient message retrieval Requirements: - The system should support more than 50 lines or extensions Ideal Skills and Experience: - Experience in developing Cloud-based PBX systems - Proficiency in integrating call recording functionality - Knowledge of setting up Auto Attendant and Voicemail to Email features - Ability to handle a large number of lines or extensions If you have the necessary expertise in building Cloud-based PBX systems with call recording capabilities and can handle a high volume of lines or extensions, please submit your...
I am looking for a freelancer who can help me configure my Grand Stream UCM 6304 IP PBX. I currently have both analogue lines and IP extensions in my setup. I require a specific configuration for my IP PBX and I am looking for someone who can provide seamless connectivity. Ideal skills and experience for this job include: - Proficiency in configuring IP PBX systems, specifically the Grand Stream UCM 6304 - Knowledge of both analogue lines and IP extensions - Experience in providing seamless connectivity for IP PBX systems If you have expertise in this area and can help me with this project, please submit your proposal. Thank you.
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.
...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 handle high call volumes efficiently, ...
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 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 project further.
I am looking for a skilled designer to create a logo for my new business. Company name is AJ Telecom we are into sales and services of all kind of telecom products like EPABX, IP PBX, Audio & Video systems please design and be able to bring my vision to life. The logo should be visually appealing and represent the nature of my business. Specific elements or symbols may be included, but I am open to suggestions in this regard as well.
I am looking for a freelancer who can assist with integrating 3cx PBX with Odoo. I currently have a 3cx phone system and an existing Odoo installation. The main task for this project is to integrate the two systems seamlessly. Skills and experience required: - Expertise in 3cx PBX and Odoo integration - Strong knowledge of VoIP systems and protocols - Experience with configuring and customizing 3cx and Odoo to meet specific business needs - Familiarity with CRM and ERP systems, particularly Odoo - Ability to troubleshoot and resolve any issues that may arise during the integration process The ideal freelancer will have a proven track record of successfully integrating 3cx PBX with Odoo and will be able to provide references or examples of previous work. They should a...
