Asterisk pbxmunkák
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Kamailio - Configuration TLS certificates, it'll probably 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 in need of a freelancer with strong skills and experience in PBX maintenance. Specifically, I have a digital PBX system that requires attention. The main issues that I am experiencing are related to configuration problems, as well as the need to migrate to Freepbx and develop new applications. Ideally, the freelancer will be able to complete this work within 1-2 months, though there is some flexibility with the timeframe. The ideal freelancer will have experience with digital PBX systems and be able to troubleshoot and solve configuration problems. Additionally, experience with Freepbx migration and application development would be highly valued.
Our company is currently in need of a skilled freelancer to assist us with our PBX system maintenance. We currently have a VoIP system in place and require maintenance to migrate the system and develop and provide solutions for repairing any issues. Ideal Skills and Experience: - Experience in VoIP systems - Knowledge in system migration and development - Troubleshooting and repair skills Timeline: - Flexible (2+ weeks) If you possess the necessary skills and experience, please do not hesitate to apply.
I am experiencing issues with my PBX and Voip systems. Specifically, I'm having connectivity issues. I am using multiple services/platforms for my Voip but I'm not sure which solutions or troubleshooting methods I should use in order to fix my current issues. I'm in need ofa freelancer to fix the issues remotely.
I need a webrtc to sip gateway to be implemented so I can connected some webrtc softphones (asterisk webrtc softphones on our Odoo CRM) to twilio sip domain
I am looking for an experienced Asterisk API developer to create an Asterisk Webhook API. The developer should have extensive expertise with Asterisk and be able to develop the API using an open source programming language. The API must include functionality for call control, call recording, and voicemail management. We are open to suggestions for the programming language and frameworks used to create the API, however, it must be versatile and reliable. We expect comprehensive documentation for the API as well as ongoing support to ensure it functions properly. Thank you for your time and we look forward to your proposal.
I need an asterisk device on my server with binding to sip number. all on my own crm laravel. I want to be able to create a web client so that you can make calls from anywhere.
Tengo varios requerimientos, comencemos por el siguiente, estando los agentes en llamada con predictivo, los comienzo a monitorear con ChanSpy(Agent/${EXTEN:-4},d), el requerimiento es que grabe la voz de quien esta monitoreando.
Installation and configuratoin of Asterisk/FreePBX on Ubuntu server Integration and configuration of softphone client on Raspberry Pi, this client will be command line based for easy integration with 3rd party API
I am looking for a freelance IOS developer to create a SIP phone app for my existing business. The app needs to connect to my FusionPBX server, which is already set up. My ideal candidate would have experience in developing IOS SIP phone apps and be familiar with the FusionPBX system. The user interface customization required is moderate. I would like the app to have shared c...required is moderate. I would like the app to have shared call appearances feature and a clean, user-friendly design. Key requirements: - IOS SIP phone app development experience - Familiarity with FusionPBX system - Moderate level of user interface customization - SHARED CALL APPERANCES - MULTIPLE REGISTRATIONS/EXTENSIONS HANDLED IN THE APP - Clean and user-friendly design. - FREESWITCH AND FUSION PBX E...
I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode
Hello, I am looking for an experienced freelancer to create a virtual phone call center using Asterisk and outbound and inbound calling services. Our organization needs a reliable, efficient and cost-effective solution to provide customer support services. The ideal candidate should have extensive experience with these technologies and must demonstrate that capability through the workload provided. Examples of previous work will be an asset. We will need them to provide a detailed proposal of your plan for the project. This system must support outbound and inbound call services, ensuring that our customers always have a friendly voice ready to answer their queries. We need a reliable and easy to use system that can handle a variety of inbound and outbound calling features such as:...
I am looking for an experienced AVAYA IP Office 500 V2 IP PBX troubleshooter who can help me with the following: - Troubleshoot the non-functional IP PBX system - Resolve the SD card problem causing extensions to not register - Provide solutions to fix the full SD card issue - Assist with creating a backup of system configuration and data - Recommend best practices for system maintenance and backup Skills and Experience Required: - Proven experience with troubleshooting AVAYA IP Office 500 V2 IP PBX systems - Expertise in resolving SD card issues and non-functional IP PBX systems - Knowledge of system maintenance and backup procedures - Familiarity with AVAYA IP Office 500 V2 IP PBX system configuration and data backup If you have the skills an...
I am seeking a FreeSWITCH or Asterisk expert to help me set up a new enterprise. The ideal freelancer should have prior experience in this area. The project requires call routing and forwarding features to be implemented. Other functionalities such as IVR and voicemail, as well as conference calling and recording may also be required. Please provide your experience in this field in your application. Automation will be required for this as well. I basically want people to go to my website, sign up, and get started with processing calls. Thank you.
We have a telephone system at We use Odoo as ERP system In order to be able to telephone directly from Odoo, we have to interpose asterisk. The goal is that we can start and accept calls from Odoo. Every employee should be reachable via their extension so that we can also use the connect function. A Linux server for installing asterisk is available. Access via ssh key
I am seeking an expert for the installation and operation of a VoIP system using Asterisk and A2Billing. The work will involve the following: Full installation and configuration of Asterisk and FreePBX on an AWS Ubuntu server. Full installation and configuration of A2Billing for managing customer accounts, billing, and prepaid services. Configuration of GSM gateways to work with Asterisk / FreePBX. Configuration of a mobile phone application to work with the system (such as Zoiper, Linphone, or Bria). Ensuring that customers can easily download the app and set up their accounts. Ensuring that the system is secure and regularly updated. I am looking for someone with prior experience working with Asterisk and A2Billing, as well as configuring GSM gateways. You sh...
...for a skilled freelancer to help me set up and integrate Asterisk PBX with our VoIP phone system and a CRM platform of my choice. Phone System: My current phone system is VoIP. CRM Platform: Although I do not have a preferred platform, I am looking for a freelancer who is experienced with integrating Asterisk PBX with different CRM platforms. Please let me know which platforms you are experienced with and your recommendations for my business. Project Timeline: I do not have a strict deadline for this project, so I am looking for a freelancer who is able to work on it with flexibility and deliver quality work. Ideal Skills and Experience: - Expertise in Asterisk PBX setup and configuration - Experience in integrating Asterisk PB...
I am looking for a freelancer to set up an asterisk PBX and integrate it with our CRM, which is EspoCRM. Our company is small, with less than 50 employees. Our main goal for this integration is to improve our customer service. Ideal skills and experience for this job include: - Experience with asterisk PBX setup and integration with CRM systems - Familiarity with EspoCRM - Knowledge of customer service best practices - Strong communication skills to ensure smooth integration and training of employees.
Hi there, I have Twilio and VitalPBX (Asterisk server). I need the following flows to be created: 1- An IVR flow via Twilio studio • 1 for Sale • 2 for Technical Support • 3 for Order follow up - After any of the above extensions are dialled, the call needs to be forwarded to the corresponding ring group of Asterisk extensions. If the dialler doesn't dial any extention, the call will be automatically forwarded after a cetrain time to operator ring groups. - At any time a caller can request a call-back by dialling a number. - Extensions can define a number (e.g. their mobile number) and have all outgoing and incoming calls originated/terminated to/from their mobile phone 2- After a call is finished, the call transcript and call summary (by calling chat...
We need to develop a SIP to WAPP gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WAPP to complete the call to the called party number. The development ...correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WAPP gateway 2)WAPP gateway converts the sip/iax signal to WAPP protocol 3) the termination number carried from the origination header will be checked by the Asterisk gateway , if the number is used by WAPP and if the number is online, the call will terminated on WAPP!. 4) if the number is not used in WAPP it sends 503 error and rerouted to ot...
Melhorar o desempenho da classificação de chamadas no Asterisk
I am looking for a person who can install and configure a SIP client(preferably PJSIP or any other) on my Raspberry Pi board, then install a Asterisk Server on a Linux computer that is on the same nextwork. After installation, configure both to talk to a VoIP SIP phone which is on the same network. Also confgigure the PJSIP client in listen only mode
I am looking for an experienced animator to help me create a promotional video for my product. My main goal is to build brand awareness and increase visibility. I would prefer a 2D animation style, and the ideal length of the video should be 30 seconds. Skills and experience required: - Experience in 2D animation - Ability to create engaging and visually appealing animations - Understandin...experience required: - Experience in 2D animation - Ability to create engaging and visually appealing animations - Understanding of brand identity and how to effectively communicate it through animation - Ability to work collaboratively with me to bring my vision to life. Reference video:
We are seeking an experienced SIPROXD server configuration specialist to assist with the setup and configuration of our SIPROXD server. Our server is hosted at IP address 4.4.4.4, and the SIP port is set to 7090. Requirements: In-depth knowledge of SIPROXD server configuration and setup Proficiency in SIP protocols and routing Experience with SIP servers like Asterisk or other SIP engines Ability to test and troubleshoot configurations Familiarity with virtual machine environments (VMs) Strong communication skills and ability to work collaboratively Job Responsibilities: Configure SIPROXD server to handle incoming calls from IP address 5.5.5.5 Route incoming calls from 5.5.5.5 to a separate SIP server at IP address Ensure proper routing
looking for a FusionPBX expert familiar with the front end and backend to setup provioning at the domain level so system will log onto device manufacturer RPS with our credentials, add MAC and Domain name info to RPS auto matically based on the MAC being added to the Fusion extension detail>for yealink, grandstream, htek and snom. *Must be familiar with modifying handset templates. Load provioning editor. Add a timer to the "Provioning enabled" that will disable after 30 minutes Setup fax availability. Check SMTP outbound setup Setup Lets encrypt for multi domain "Sub-domains" with a viewable page from the Pages to be added to the front end; to show currentLets Encrypt certs and details like expiry. Firewall to show and edit blacklist and whitelisted IP's...
I am looking for a freelancer to create or sell an open source CentOS based IVR application that will have advanced IVR functions. The ideal candidate should have experience with PHP programming language. The IVR application should be able to handle more than 50 users interacting with it. The following features are requir...looking for a freelancer to create or sell an open source CentOS based IVR application that will have advanced IVR functions. The ideal candidate should have experience with PHP programming language. The IVR application should be able to handle more than 50 users interacting with it. The following features are required: - Advanced IVR functions - Customized IVR functions - Basic IVR functions - Asterisk Based - Should Also Install Application And Give Installati...
Integration of Asterisk FreePBX into Perfex CRM Hello, We are seeking a skilled freelancer to assist us in integrating Asterisk FreePBX into our Perfex CRM system. Our goal is to streamline communication processes and enhance our customer relationship management capabilities. Please note that we have some additional requirements to add to the initial project description. Here is the complete list of requirements: 1. Expertise in Asterisk FreePBX: You should have a strong understanding of Asterisk FreePBX and its features, such as call routing, voicemail, IVR, call recording, and reporting. 2. Proficiency in Perfex CRM: Familiarity with Perfex CRM or similar CRM systems is essential. You should have experience working with APIs and integrating third-party too...
I am looking for an experienced freelancer to develop an IVR system in Asterisk for my customer service needs. The project will involve creating IVR prompts based on a pre-existing script provided by me. The system should have 6-10 menu options for callers to choose from. Ideal skills and experience for the job include: - Strong understanding of Asterisk and IVR systems - Experience in creating IVR prompts and scripts - Ability to customize IVR menus based on specific business needs - Familiarity with integrating IVR systems with other business systems and applications If you have the required skills and experience, please submit your proposal with relevant examples of past work. I look forward to working with you.
hi i have issabel pbx and i want to integrate it woth zoho crm to send and recive calls from zoho crm
I am looking for someone who can help me with a Yeastar PBX API data push to Postman project. Specifically, I need to push call logs, call reports, and voicemail messages...etc, to Postman using JSON. I have access to the Yeastar PBX API documentation and my preferred programming language for this project is JSON. Ideal skills and experience for this job include: - Strong knowledge of Yeastar PBX API - Proficiency in JSON - Familiarity with Postman - Ability to work efficiently and meet deadlines.
I need someone to connect my Twilio account to my Yeastar Cloud PBX via Sip Trunk. The main purpose of this project is to make and receive calls. Currently, I have a Twilio account, but I require guidance on best practices for the SIP Trunk configuration. The ideal candidate should have experience with Twilio, Yeastar Cloud PBX, and SIP Trunking. The project is urgent, and I need someone who can complete the task quickly and efficiently.
I'm looking for professional support with Issabel PBX set up, and am open to either a one-time job or an ongoing relationship. In addition, I have specific requirements for the job that I am looking for someone experienced to manage. This is a remote support job opportunity that requires expertise in PBX setup and customization of configurations to fit the needs of our organization. If you have experience in this field, I want to hear from you. We have a clear setup requirements total of 12 Sip trunks with 4 different providers. We will supply the sip credentials. we want inbound calls and outbound calls to be direct on to the PBX with IVR defined. we have already made 3 mp3 files for IVR.
We are looking a telecom engineer to configuration and manage a powersmpp sms platform (need for sms transit) and goip devices on a part time basis. need know softwares powersmpp, asterisk, easyphone. thank you
Call Forwarding Portal with CC and Call Limit I am looking for a freelancer who can create a web-based portal for call forwarding with simultaneous ringing. The ideal candidate should have experience in developing call forwarding portals and be proficient in Asterisk. Features: - Simultaneous ringing - Call forwarding with CC and call limit - Customizable settings - CRM integration with Asterisk - Analytics and reporting Skills: - Proficient in Asterisk - Experience in developing call forwarding portals If you have the necessary skills and experience, please submit your proposal. Thank you.
I would like a multi-class Asterisk tutorial, at first with the most basic concepts (from zero) and increasing the level with each class, starting to beginners and finishing with a professional mastering of the subject, to learn and understand completely how to use Asterisk and become proficient of the subject. The minimum total duration of the course with all classes must at least 8 hours. I need good quality on the recording and the edition of the course. Also, the content has to be original, to be able to use it without restrictions. We will sign a contract of copyrighting cession before ending the project.
Hello guys i have successfully installed fusion pbx on my aws account . But its not getting connected with external softphone look like some IP issues please contact me
Hello, i use this software from github. My settings are: Outgoing calls use extension 333 which forces calls to "optional destination" to IVR. (After uploaded contact list, script start calls and connects to the IVR.) The problem is: Have current settings for outgoing calls via TRUNK (DongleX). I want to be able to c...is instaled in freepbx and it works normally. $callFile = "Channel: Dongle/dongle1/$numbern"; $callFile .= "CallerID: $caller_idn"; $callFile .= "Context: callblastern"; $callFile .= "Extension: 333n"; Task: Set outgoing connection via another TRUNK (not Dongle). (Missing - correct outgoing "context" and setting in FreePBX) I am looking for someone with experience. freepbx 15, asterisk 16 r...
Job Description: Have installed the latest version of FreePBX on new hardware and Cisco SPA8000 . Requiring a talented Freelancer for this system to: Connect to Yealink T28P x 8 Connect to 2 physical PSTN phone lines 1 Fax line via Cisco SPA8000 Route calls from Cisco SPA8000 to FreePBX A separate inbound queue for each phone line Outbound line selection based on gro...Yealink T28P x 8 Connect to 2 physical PSTN phone lines 1 Fax line via Cisco SPA8000 Route calls from Cisco SPA8000 to FreePBX A separate inbound queue for each phone line Outbound line selection based on grouping Configure outbound email notification! Headset programming Set up / Configure Sangoma Softphone App Any other minor tweaks. Immediate start 5/1/2023 10:00am EST! Skills: Asterisk PBX,...
I am looking for a freelancer who can assist me with a Ubuntu Asterisk SIP problem. My current setup involves Ubuntu 20.04 and Asterisk 18. I am experiencing a general issue with outgoing calls being masked, as the number in Zoiper is not being recognized. Ideal Skills and Experience: - Strong knowledge of Ubuntu and Asterisk - Experience with SIP configurations and troubleshooting - Familiarity with Zoiper and other SIP clients - Ability to identify and resolve connectivity issues - Experience with implementing advanced features such as call recording and IVR systems Overall, I am seeking a skilled freelancer who can help me identify and resolve the issue with my outgoing calls, and ensure that my SIP configuration is optimized for call quality and reliability.
Hi there, I'm looking for an experienced freelancer to help me with integrating an existing VoIP platform with an existing CRM. Voice over internet protocol(VoIP) is an ever-evolving technology that can greatly improve the way businesses communicate both externally and internally. I will be using FreePBX/Asterisk as the VoIP platform, and amoCRM/Kommo as my Customer Relationship Management (CRM) system. I need the following integration: VoIP Integration to allow real-time communication via telephone, Sales Pipeline Integration to track sales leads in my CRM,Click2Call, Incoming Call Card, Creating a Contact card, Smart Forwarding, Call Results card, Call Logging, Incoming Leads, Call List, Built-in call feature (WebRTC). All of which has examples on
...d'un freelancer pour m'aider à configurer Kamailio sur mon serveur et orchestrer plusieurs serveurs Asterisk. Mon objectif est de recevoir des appels de Twilio et Telnyx, et de mettre en place des règles entrantes et sortantes pour gérer ces appels. J'ai déjà installé un docker de Kamailio sur mon serveur, mais je ne sais pas comment configurer les premiers réglages. J'aimerais utiliser le mode file plutôt que la base de données, ce qui signifie que j'aurai un fichier de réglages contenant les adresses IP de mes serveurs, les adresses de Twilio et Telnyx, ainsi que les règles entrantes et sortantes pour gérer les appels. Je suis à la recherche d'un freelancer ...
I have 8 FXO and only accessiable via SIP. They are BT1, BT2, BT3, BT4, BT5, BT6, BT7 and BT8 I need asterisk to check for call counter before making a call. Only least / lowest number of the BT port counter can be dial out. Below is my logic. Everytime when asterisk receive a dial request 1. asterisk will check which BTs has the least number of call 2. once known, asterisk will dial the call. for example BT1 3. Regradless of the status of call, (ANSWERED, BUSY. UNAVAILABLE), asterisk will add BT1+1 The idea is to evenly distribute the call between all FXOs. You will need to have anydesk and access to my PC to work as asterisk is only available locally. (LAN)
Hello: We have 2 portals with same origin (PBX system) for calls and faxes. We want to make UI/UX better by converting raw PHP into Laravel. You will make new landing pages with modern Dashboard. You DO NOT need to make any PBX settings or changes. Our PBX specialist can help explain the UI if needed. I am sure an ambitious programmer can make it much better in a reasonable budget. We are open to use a program other than Laravel as long as our raw PHP code will remain the same. Please bring your proposals and suggestions. Thank you
Required knowledge of Linux (CentOS), Asterisk & FreePBX. -I need a command to delete files older than 32 days in the Asterisk recording folder and voice mails. /var/spool/asterisk/voicemail/default/ => file pattern msg????.???, recursive /var/spool/asterisk/monitor =>all files, recursive -I need to create a cron job to execute the deletion weekly. The user (meladmin) I use to manage my server is in the following groups: meladmin, wheel and asterisk. But I cannot modify or delete files created by Asterisk. -I need to be able to modify and delete asterisk created files via WinSCP. How do I modify the meladmin user? -I’m trying to use the Voicemail blast. I can call it from an IVR, but the extension cannot access it. How to...
solve audio problème on Issabel PBX and configure dialer plugin
I am looking for someone who can help me set up a UK Australia toll-free system. i need sip credential that primarily focus on inbound and outbound. I would like the toll-free service to have call recording capabilities as an additional feature. The anticipated call volume is low, Ideal skills and experience for the job include: - Experience setting up toll-free services - ...toll-free system. i need sip credential that primarily focus on inbound and outbound. I would like the toll-free service to have call recording capabilities as an additional feature. The anticipated call volume is low, Ideal skills and experience for the job include: - Experience setting up toll-free services - Knowledge of call recording software - Have skill on open source PBX system like Fusion pbx...
I am looking for a freelancer to integrate Cliniko CRM and Yeastar PBX for my business. The main goal of this integration is to have an inbound caller ID popup based on existing customer's caller ID. The Yeastar PBX system is already set up and configured, so no additional work is needed there. I do not require any additional features besides the caller ID popup. Yeastar API Cliniko API Ideal skills and experience for this job include: - Proficiency in Cliniko CRM and Yeastar PBX - Experience with call tracking and logging - Knowledge of VoIP systems - Attention to detail and ability to troubleshoot any issues that may arise during the integration process
I'm in need of a configuration for an Issabel PBX installed on an AWS EC2 instance, with local compatible gateways connected. The configuration needed is the basic setup, and I do have compatible gateways available. This setup will be using a single network topology.
I am looking for a freelancer to connect my PBX to WhatsApp. Specifically, I am looking for a SIP trunk connection. The successful freelancer should demonstrate strong experience in this field, and provide detailed project proposals with their application. I have a budget of $100-2,00 for this project.
