A2billing freepbx asteriskmunkák
I am looking for a freelancer to set up Kamallio as ...freelancer to set up Kamallio as a load balancer for inbound calls. This project is for an existing business and we require a round-robin algorithm for load distribution. We expect a call volume of more than 500 per day. In addition, we would like to have reporting features for analytics. A deployment and demonstration with all code edits of example application like from here - transcription app will be good. Ideal skills and experience for this job include: - Experience in setting up Kamallio as a load balancer - Knowledge of round-robin algorithms for load distribution - Familiarity with handling high call volumes - Ability to integrate reporting features for analytics
I am looking for a skilled developer who can help me with the development of a FreeSwitch + API VoIP system. The main functionality that I need for my system is call routing and forwarding. I do not have an existing VoIP system that needs to be integrated with FreeSwitch. For the API development, I am open to suggestions regarding the programming language to be used. Ideal skills and experience for this project include: - Strong knowledge and experience with FreeSwitch and VoIP systems - Proficiency in API development - Flexibility and ability to suggest the most suitable programming language for the API development - Familiarity with CRM integration would be a plus
Overview: This project aims to build an AI-powered voicemail system using either Twilio or Asterisk. The system will utilize advanced technologies like voice recognition, text-to-speech, and intelligent call processing to enhance the voicemail experience for Restaurant Baguette. Objectives: 1. To develop an interactive voicemail system with intelligent call routing and response capabilities. 2. Integration of voice recognition and text-to-speech technologies 3. To provide detailed documentation for system setup and troubleshooting. Call Flow: 1. Call answered on Twilio DID (or Asterisk vm). 2. Play recording: “You’ve reached the smart voicemail of Restaurant Baguette, how can I help?” 3. Forward call to “Vocode” () 4. Set up Vocode with Deep...
Hello, I want to work with WhatsApp, the solution should be a web page application based on sip+webrtc.1: User A make a call to whatsapp 2: Wahatsapp receive a text message 3: User A receive ringback 4: Whatsapp click on link and establish the video call are you able to do something like that Regards
...I'm looking for a knowledgeable individual based in the UK who can assist me in starting a VoIP service provider venture. I need guidance and expertise in setting up the infrastructure and initiating operations within the Asterisk PBX framework. **Key Deliverables:** - Consultation and guidance on establishing a VoIP service provider setup tailored to the UK market. - Configuration and setup guidance specifically within the Asterisk PBX framework. **Submission Requirements:** - Overview of previous experience in setting up VoIP service provider infrastructure or Asterisk PBX systems. - Proposed plan or strategy to assist in initiating the VoIP service provider setup. - Estimated timeline and pricing structure for consultation and support services. **Note t...
I am looking for a freelancer who can install Asterisk on my Ubuntu 20.04 LTS server for VoIP services. I have specific requirements for the configuration and customization of the Asterisk installation. Skills and Experience Needed: - Strong knowledge and experience in installing and configuring Asterisk on Ubuntu 20.04 LTS - Expertise in setting up VoIP services with Asterisk - Ability to understand and implement specific requirements for the Asterisk installation If you have experience with Asterisk installations and can meet these requirements, please submit your proposal.
Project for Guiliano B Objective: -Setup VOIP Timeline: - Desired completion within 1-2 weeks Expertise required: - FreePBX Additional requirements: - Setup VoIP system for improved communication
I am looking for a skilled professional to help me set up a VoIP gateway with Asterisk. Integration with our own server is a must, so experience in this area is required. After the initial setup, I will require occasional troubleshooting support. Ideal skills and experience for this job include: - Strong knowledge of VoIP Gateways and Asterisk - Experience with server integration - Ability to provide occasional troubleshooting support
I am seeking a freelancer who can provide one-on-one lessons in VOIP, PBX, and Asterisk. As a beginner in this field, I am looking for guidance in setting up a PBX system, configuring Asterisk, and troubleshooting VOIP issues. The ideal freelancer should have advanced knowledge in these topics and be able to provide theory support and practical configuration instructions.
...manager with experience in Asterisk PBX systems to help me set up and maintain my communication infrastructure. Specific features needed: - Call routing and forwarding to ensure efficient call management - Voicemail and greetings for professional and personalized customer interactions - Conference calling for team collaboration and remote meetings Integration with other communication tools: - Integration with other communication tools is required to streamline communication processes and ensure seamless connectivity. Ongoing maintenance and updates: - Assistance with regular maintenance and updates of the Asterisk PBX system is needed to ensure smooth and uninterrupted communication. Ideal skills and experience: - Strong knowledge and experience with Asterisk P...
Hello, I recently acquired an OpenVox SWG-3032 and am looking to set up a telephony system from scratch for my office. Here are my basic requirements, more functionalities are welcome: 1. **Gateway Model**: OpenVox SWG-303...calls) - Outbound IVR calls for customer feedback, including detailed analytics (call accepted/rejected/no answer/answered but no input, etc.). 4. **Infrastructure**: We have a PC acting as a server, a NAS with Docker support (Unraid), and a static IP. 5. **Timeline**: As soon as possible. Could you please guide us on setting this up, and suggest a suitable CRM that integrates well with 3CX or FreePBX or Asterix or any other? Also, what would be the estimated timeline and other details for this project? Looking forward to your expertise and assistance. Be...
I am looking for a VoIP developer who can create a VoIP application for my project. The ideal candidate should have experience in developing VoIP applications with the following functionalities: - Call routing and forwarding - Call recording and monitoring - Voice recognition and transcription I have no specific requirements or preferences for the programming language to be used, so the developer can choose the most suitable language for the project. The VoIP application is expected to be used by: - Less than 100 users - 100-500 users - More than 500 users If you have experience in VoIP development and can fulfill the above requirements, please submit your proposal.
I am looking for a freelancer who can help me with a project involving an asterisk based system like free pbx. The specific features I need for the system include call forwarding, voicemail, and IVR (Interactive Voice Response). I have no specific requirements or preferences for the system, and I am open to suggestions from the freelancer. The deadline for this project is within a week. Ideal Skills and Experience: - Strong knowledge and experience with asterisk based systems - Proficiency in configuring call forwarding, voicemail, and IVR - Ability to suggest and implement best practices for the system - Excellent problem-solving skills and attention to detail - Timely delivery of project within the specified deadline
WordPress module. I need a module that will serve as an employee database added by companies. Due to GDPR, the entry in the database should look like this: 1. Name and surname (only the name should be shown on the website and the surname, the first letter and the rest under the asterisk are not visible). Here we choose a female/male avatar. 2. ID number consisting of 11 digits on the website we see only the first 2 and last 2 digits, the rest under an asterisk 3. City 4. The job position he held 5. Duration of employment 6. Rating on a scale of 1-10 stars 7. Employer's opinion - a place where we can write a subjective opinion. 8. Reason for terminating the contract: by the employee, by the employer, by mutual consent of the parties. The module requires a search engine w...
...Trunk user with PBX support I am looking for a skilled professional who can help me with the remote installation of a PBX system. The ideal candidate should have experience working with Asterisk PBX. Requirements: - Familiarity with Asterisk PBX system - Ability to remotely install and configure the PBX system - Proficiency in setting up new features for the PBX system - Experience in configuring voicemail services - Knowledge of call forwarding and interactive voice response (IVR) setup Skills and Experience: - Previous experience in installing and configuring PBX systems - Strong knowledge of Asterisk PBX system - Ability to troubleshoot and resolve any issues that may arise during the installation process - Proficiency in setting up voicemail services and conf...
I am looking for a freelancer who can integrate our Odoo community with our Asterisk-Issabel Phone Center. The specific feature that we want to integrate is the ability to be able to make calls straight from Odoo. We are open to suggestions for the preferred method of integration. The timeline for this project is immediate, with a deadline of 1-2 weeks. Ideal skills and experience for this job include: - Strong knowledge and experience with Odoo community and Asterisk-Issabel Phone Center - Proficiency in CRM integration - Familiarity with different integration methods, such as Direct API and Web Services
i want to implement a simple architecture. at this moment my asterisk server works fine with endpoints and it can initiate calls in both directions. now i want to use a proxy server in between endpoints and my server . so im using dsiprouter with domain name of and its installed successfully. now there is a problem. i connect my sip users to dsiprouter. and dsiprouter to asterisk as pass thru. when i call any number with sip endpoint there is no problem and the call reaches the dinstar gateway and dinstar call the sim number(ip2tel). but the opposite side of call when sim user call the gateway and gateway route the call through asterisk. when asterisk send request to dsiprouter the request cannot find the endpoint i tried (realm-outbound proxy and etc) but i di...
I am looking for a freelancer who can help me with the installation of VitalPBX 4 (based on asterisk) connected with a Twilio number. OS is Debian. Additionally, I require complete configuration of this addon: (it has a .sh script auto-installation) After all is done, YOU MUST GUIDE ME to make the first succesful call with the addon.
Project Description: I am looking for a skilled developer who can create a WhatsApp to SIP Gateway using Asterisk. The main function of this project is to enable call forwarding between WhatsApp and SIP. I require the gateway to be built on an open-source platform. 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. Skills and Experience Needed: - Strong experience with Asterisk and VoIP systems - Proficiency in working with both WhatsApp and SIP protocols - Knowledge of call forwarding and routing techniques - Familiarity with open-source platforms for building gateways - Ability to troubleshoot and debug any issues that may arise during the development process
...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...
I am looking for an expert freelancer to integrate SIP services with med...freelancer to integrate SIP services with mediasoup for my project. The purpose of this integration is to enhance the video calling capabilities on my application. Requirements for the project: - Experience: The freelancer should have expert level experience in SIP video integration with mediasoup This Project may require expertise in VoIP technologies along with WebRTC, Asterisk, kamailio etc. - Tools and Technologies: NodeJS is preferred, however to get the project integration we are open to suggestions and do not have any specific tools or technologies in mind. If you have the required expertise and can provide guidance on the best tools and technologies to use, please bid on this project.
I need to install a2billing + freepbx on my server. Create a SIP account and configure a trunk. It must be proven that it works. within 24 hours.
I have vicidial and asterisk installed, I need the following and the necessary configurations in vicidial for this to work. 1.- Create 17 trunks that are in the local network, [15 goip + 2 skyline gateway] 2.- Generate a way to call certain trunks together, easily, hopefully with just a few clicks: Example 1: call trunk 8, number 103 and play the dtmf tone of * Example 2: call trunk 9, number 103 and play the dtmf tone of * Example 3: call trunk 10, number 103 and play the dtmf tone of * Example 4: call trunk 16, number 103 and play the dtmf tone of * Example 5: call trunk 17, number 103 and play the dtmf tone of * Example 6: call all trunks 1, 2, 3, 4, 5, 6, 7, 11, 12, 13, 14 and 15 together, to number 103 and play the dtmf...
I am looking for a freelancer who can configure Asterisk freePBX with Linphone client for ZRTP encryption to enable secure VoIP communication. Here are the requirements for the project: - The primary purpose of the configuration is to enable post quantum encryption for secure VoIP communication. - The specific version of Asterisk and Linphone is not specified, any version will do. - There will be no requirement for ongoing maintenance or updates after the initial configuration. Ideal skills and experience for the job: - Strong knowledge and experience in configuring Asterisk freePBX and Linphone. - Familiarity with ZRTP encryption for secure VoIP communication. - Ability to work independently and deliver the project within the specified timeline. If y...
I am looking for a freelancer who can help me restore a FreePBX on a new Contabo VPS server. Here are the details of the project: Version of FreePBX: 14 - The old server has FreePBX version 14 installed. Backup of old FreePBX server: Yes - I have a backup of the old FreePBX server. Preferred VPS server: Contabo VPS - I prefer to set up the new FreePBX on a Contabo VPS server. Skills and Experience: - Experience with FreePBX restoration and migration. - Proficiency in working with Contabo VPS server. - Knowledge of FreePBX version 14. restore all configuration of old server If you have the necessary skills and experience, please bid on this project.
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between F...
Connect FreePBX with DIDWW and Extensions, it was working but today i receive a 401 authentication failed in didwww Skills and Experience Needed: - Experience with FreePBX and Asterisk - Proficiency in setting up and configuring SIP trunks - Knowledge of connecting and configuring extensions in FreePBX Project Details: - I am using FreePBX as my PBX system and I need to connect it with DIDWW. - I prefer to use a SIP trunk for the DIDWW connection. - I have 1-5 extensions that need to be connected. - The freelancer should have experience in setting up and configuring SIP trunks in FreePBX. - They should also have knowledge of connecting and configuring extensions in FreePBX. - The project involves ensuring a seamless connection between F...
I am seeking a freelancer to assist me with setting up my issabel PBX system for incoming and outgoing call recording. Requirements: - Experience with issabel PBX system - Knowledge of Asterisk, FreePBX, and Elastix - Ability to configure the PBX system for call recording - Familiarity with telemarketing purposes for PBX setup Tasks: - Set up the issabel PBX system for incoming and outgoing call recording - agent call out with display random cli, if customer missed this call and return a call will back to particular agent. - Configure the necessary settings for seamless call record - Ensure compatibility with our existing hardware and software - Provide guidance and recommendations for optimal telemarketing functionality If you have the required skills and experience in ...
**Job Title:** Kamailio & Asterisk VoIP Solutions Architect **Job Description:** We are seeking an experienced Kamailio & Asterisk VoIP Solutions Architect to design and implement a comprehensive telecommunication solution that integrates WebRTC, SIP users, and robust VoIP services. The ideal candidate will have extensive knowledge in setting up and configuring multi-tenant VoIP systems, ensuring high availability, and developing secure and scalable APIs for system interactions. **Key Responsibilities:** - Design and deploy a multi-tenant VoIP solution using Kamailio for SIP routing and trunk configurations, and Asterisk for media processing. - Implement WebRTC functionality for browser-based communication, alongside traditional SIP user capabilities. - Con...
...freelancer who can help me integrate my VoIP Asterisk phone system with Google Sheets. I want to track caller ID and call details in real-time. I've attached an example of that is required, the phone system is Yeastar P-Series I will provide the Asterisk manager interface details Skills and experience needed: - Proficiency in Asterisk and VoIP systems - Experience with Google Sheets API - Knowledge of real-time data integration - Strong problem-solving skills to troubleshoot any potential issues Project requirements: - Connect the VoIP Asterisk phone system to Google Sheets - Set up a live CDR feed to track caller ID and call details - Ensure real-time updates of the data in Google Sheets Please note that there are 1-5 phone lines connected to the Vo...
I need a freelancer who can troubleshoot and repair my FREEPBX system as I am unable to make outbound calls. Although there are no specific error messages, there are unusual system behaviors. I have not made any recent changes or updates to the system. Skills and experience required for this project include: - Expertise in FREEPBX system troubleshooting and repair - Knowledge of VoIP protocols and configurations - Familiarity with Asterisk and Linux operating systems - Ability to diagnose and resolve issues with outbound calls - Strong problem-solving and communication skills The freelancer must be available to start working on the project today.
...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...
Necesito instalar freepbx + a2billing + 2 troncales en red local y 2 usuarios para llamar.
Linux / Asterisk server with 8 port sim gateway outbound call center. (partially inbound) agents with ip phone/gsm phones Agents must able to hold/CB/transfer call to supervisor and other functions with key buttons only. other all standard features. daily excel report for whatsapp pls quote.
Looking for a FreePBX Expert that can help configure my instance. Also I have this extension for my CRM which I need assistant with connecting to my instance and configuring.
...hardware. The requirements for this project are as follows: Operating System: - The VOIP server should be installed on a Linux operating system. VOIP Software: - The specific VOIP software that should be used for the PBX installation is Asterisk. Features and Functions: - The VOIP server should have the capability of call forwarding. - No other specific features or functions are required at this time. Ideal Skills and Experience: - Experience in installing and configuring VOIP servers. - Proficiency in working with Linux operating systems. - Familiarity with Asterisk VOIP software. Please provide your relevant experience and any certifications you may have in this area. Additionally, if you have any suggestions or recommendations for the hardware or software configura...
Develop a freePBX module with the following functionalities for utilizing my chan dongle to send and receive SMS: 1. Establish an Incoming SMS Portal 2. Enable SMS Sending 3. Facilitate USSD Sending 4. Provide Balance Checking 5. WEBRTC ( Call over web) 6. Support Dongle Setup and Configuration
This is the work that need to be done: 1. you will setup a Freepbx on a provided with any desk connection. 2. you will setup the SIP Trunk for incoming and out coming calls. (I provide you all the setting data) 3. you will setup the IVR. (simple IVR with different message for opening time and closing time) 4. you will setup 5 extensions 5. you will configure 3 Cisco Phones with the FreePBX, the phone need to be registered and tested for incoming calls, out coming calls, hold, transfer call. 6. you will setup the pfsense with freepbx rules 7. We together do a full test of the system to be sure that is all working.
I need to setup a new freepbx machine, 1 SIP TRUNK, 3 Cisco SIP Phone (You need to be specialist at setting this), 1 IVR.
I am looking for a freelancer who can help me set up FreePbx/Asterisk and provision 4 SIP phones for my VoIP phone system. Current Phone System Setup: VoIP phone system Software Installation: I already have the FreePbx and Asterisk software installed. SIP Phones: I am using Cisco SIP phones. 2xCP-6851-3PCC Phones 1xSPA-303 Phone 1xGigaset C530A Skills and Experience: - Experience with setting up FreePbx and Asterisk software - Knowledge of provisioning SIP phones, specifically Cisco phones - Familiarity with VoIP phone systems and configurations I NEED AN EXPERT IN CISCO PHONE.
Magnus Asterisk Inbound and Outbound DID Setup Skills and Experience Required: - Proficiency in Asterisk setup and configuration - Experience in setting up both inbound and outbound calling - Familiarity with DID providers and integration - Ability to customize Asterisk setup at a basic level Project Description: We are looking for a freelancer who can help us set up an Asterisk system with both inbound and outbound calling capabilities. We already have a DID provider in place and require basic customization for our Asterisk setup. Tasks: - Configure Asterisk for inbound and outbound calling - Integrate our existing DID provider with the Asterisk system - Customize the setup at a basic level to meet our requirements If you have experienc...
configurar telefonos ip cisco en entornos asterisk sin necesidad CEM CUM. instalacion de firmware y carga de configuraciones mediante tftp
I am looking for a freelancer who can assist me with setting up a freepbx server. I already have all the necessary hardware and software in place. I need the freelancer to configure asterisk to meet my business needs and migrate my existing PBX to Alibaba Cloud. The ideal candidate should have experience in freepbx, asterisk, and Alibaba Cloud. The project should be completed within a week.
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" .
Please help me connect JIO sip to my freepbx server... Ping ia working perfectly.. Only outgoing is not going to any number thiugh numbers are running and incoming is also not working..
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...
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.
...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...
