Sphinx asterisk jobs
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.
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 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 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.
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 PBX with CRM platforms suc...
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...
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
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
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
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, 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 ...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 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, Linux,...
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
...require a high level of detail and realism in the final output. We have specific software and file format requirements for the final output, which will be shared with the selected freelancer. Ideal Skills and Experience: - Experience in VFX and 3D animation - Expertise in high-poly/photorealistic animation - Proficiency in the required software and file formats for the final output. Final brief SPHINX VFX 33,36,37,38,116a ,116b and 118 We need a number of shots to be completed:- When you send completed shots please:- 1. label them with their VFX number e.g. VFX 2 and a version number. 2. All shots must be output as MOV files using the Pro Res 4444 codec. 3. DO NOT colour grade shots unless we specifically ask you to. 4. Send a screenshot or previous very 2 da...
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)
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...
VFX Brief – WIDE SHOTS OF THE SPHINX INSIDE A CAGE HANGING ABOUT CHURCH FLOOR SHOT NUMBERS: VFX 33A, 35A, VFX 64A AND VFX 64B. We need a number of shots to be completed:- When you send completed shots please:- 1. label them with their VFX number e.g. VFX 2 and a version number. We will not accept files that are not labeled with their shot number. Thanks. 2. All shots must be output as MOV files using the Pro Res 4444 codec. 3. DO NOT colour grade shots unless we specifically ask you to. VFX 33A AND 64B - WS SPHINX IN CAGE. LOW SHOT LOOKING UP. Can you make the underneath of the base of the cage plain without the white marbling effect please. Link to the shot is here. Can you make the cage swing
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 - Knowledge ...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 or asteri...
Installation of A2Billing latest and asterisk on a debian server. For future integration with linphone SDK I want someone who has past experience in configuring asterisk to work with A2Billing. Your job will be to and install A2Billing billing 2. install and configure Asterisk with cdr to be working on A2Billing I should be able to an account from A2Billing and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in A2Billing billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation.
Hi, I have a new installation of freeside billing 4 and asterisk on a debian server. I want someone who has past experience in configuring asterisk to work with freeside. Your job will be to free side billing 2.Asterisk I should be able to an account from freeside and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in freeside billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation. Thanks
I'm lookin for of a skilled engineer to manage and deploy a SIP & Kamailio on docker compose that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engin...environment (not final, but wished): - Ubuntu 22.04 - dockerized - in a later task, to enable kubernetes for Kamailio and HA Good entry points: Budget? will not be disclosed. So place your best hourly rate!
I'm looking for help resolving an issue I am having with Asterisk on a Linux server. I'm unsure of what caused it, but it needs to be fixed urgently. The ideal freelancer for this project should have expertise in System Administration for Linux platforms. I'm open to suggestions on how to diagnose and remedy the issue, so if you think you can help, don't hesitate to reach out and make your case. We are getting a lot of chan_sip.c:4413 __sip_autodestruct and have tried restarting the server once but without any positive results.
I'm looking for experienced mentoring services to help setup and integrate an existing SBC system with Asterisk. I need assistance for 1-3 people with setting up the system, configuring SIP trunking, and ensuring that the integration with Asterisk is working properly. With the right combination of assistance, knowledge, and experience, I'm confident that this project will be a success.
I'm in need of a skilled engineer to manage and deploy a SIP & Kamailio & Asterisk solution on-premise that also requires some admin services. I need these services done on a part-time basis, but the engineer must stay available and be ready to help should any issues arise. This means it is essential that the engineer is experienced with the SIP & Kamailio & Asterisk solution and be comfortable dealing with any issues that may arise. The engineer must also have experience in deployment on docker & docker-compose and later on administration of this system. This position will require some extra work every week, but I'm ready to discuss the amount of work needed with the chosen expert in order to get the best possible results. We usually di...
Hello! I am looking for a freelancer to help me set up a VoIP system using Asterisk and a Skype trunk specifically for external calls. This setup is for my personal use, not for an existing or new business venture. The setup should be able to handle a high volume of external calls with clear voice and quality. I expect a complete solution in terms of setup and installation of all the VoIP components to ensure that I am connected and ready to place external calls. I am essentially looking for someone who can configure Asterisk and the Skype trunk with good quality, so that I can get started making external calls. Thank you!
Need to integrate Asterisk PBX into Zendesk. Using Asterisk AMI and POST API into Zendesk. When a call comes in on a specific ques ticket must be created in Zendesk using caller ID as a requester (if the customer number is not in Zendesk).
we've installed manticore but seems to be conflicting with old searchd / sphinx Need it working so that we can utilize for search on database.
I have issues with the production server so I am looking someone to fix it asap. I have 2 different applications assigned to the the same carrier and carrier transfers calls simultaneously to both the ip, but right now as the call reaches fusion its disconnecting the call not allowing it to get transferred to asterisk server.
I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded. The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface. I need to have some new phones connect to the Server over TCP Port 5062 I need to have some new phones connect to the Server over TLS Port 5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL T...
This is for a Asterisk and Zoho CRM integration. We need to receive and make calls on Zoho. Also be able to record, transfer, pop up a window with caller contact on Zoho.
API SDK module required with Asterisk using PHP and MySQLwith subscription and credit billing model for users to enable them to add their own SIP trunk and route their own sip accounts and use API through any of the SIP trunk they have to process their IVR instructions on given webhooks. Basic MySQL structure to start with. a) users table [api_key (unique), loginid (unique), pwd, trunk_id , user_type (admin/user) ] b) trunks table [trunk_id, username, password, ip/address , port, protocol, transport_type, did_numbers, status] and other fields and table that is required. Experience Required : Asterisk AGI PHP MySQL Reference API SDK documentation Calls will be made through the voip server using the rest API tht needs to be developed in PHP below is the sample how we will
Hi Pablo B., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I'm installing Kommo CRM in my company and we currently use voip offered by local telephone company. I'd like to configure our voip numbers on asterisk and then Integrate with kommo CRM. Can you do it?
I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C due to VPN. Thanks
I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:
I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.
...which will provide the pop3 email access to the phone system in order to fetch the new emails and allow the customer to control their emails by reply, forward, delete or mark as spam by selecting options on the phone keypad or by voice commands. Each customer will have an email account on the email server and will be able to access the email address by placing a call to the phone system (based on Asterisk pbx). Each customer will use an authenticate method to authenticate and allow the access to email account through the phone system. Each customer will be able to import up to 5 external email accounts (Gmail, Yahoo, AOL, ... etc) by using the POP3 access and the system will allow the customer to manage these emails with their email account at our private domain. Also, it wil...
We are having major issues with our vicidial. Channels not connecting and calls being burned and reps sitting waiting. We are losing money and employees daily and need someone to help us get dialer running smooth asap.
