Didspekerjaan

I am looking for an expert in Yeastar p560 VoIP systems to implement a solution for a trunk to trunk call forwarding inclusive of Direct Inward Dialing (DIDs). The task is troubleshooting headers, the trunks / solution is already setup. The key functionality includes: - Trunk to trunk hand off with headers (in and out) - Forwarding calls over VoIP Your expertise should span the configuration of VoIP systems for diverse and specific users. Knowledge of wireshark and pcap analysis, understanding of SIP headers. With your understanding of my needs, you'll enable seamless communication within my organization. Prior experience in similar setups is a necessity. Your bid will be a value-add to the success of this project.

...solutions is key for success in this role. Effective Communication: Clear and concise communication skills are essential for collaborating with our team and ensuring that project requirements are Stack: Blockchain Platforms: Ethereum, Hyperledger Fabric Smart Contracts: Solidity, Chaincode (GoLang) Decentralized Storage: IPFS, Storj, Sia Identity Standards: Decentralized Identifiers (DIDs), Verifiable Credentials Cryptography: Public-key cryptography, Zero-knowledge proofs Development Tools: Truffle, Remix, Ganache, Metamask Backend Development: Node.js, Frontend Development: React, Vue.js, Requirements: Proficiency in Blockchain Development: We're seeking a developer with expertise in blockchain technologies, including Ethereum, smart contracts, and decentralized

...telephony features programmatically, including customer management, DID handling, call routing, and call feature configurations. This project requires deep knowledge of FusionPBX, FreeSWITCH, and database management using MariaDB. Key Project Tasks: Customer Management: Create and update customer records, including status management (Active/Inactive). DID Management: Insert, update, delete, and route DIDs within the system. Call Feature Configuration: Enable/disable call recording for specific numbers, and set/modify caller IDs. CDR Synchronization: Develop a mechanism to synchronize Call Detail Records (CDR) from FreeSWITCH to a SQL-based application. API Security: Implement robust authentication and authorization for the API to ensure secure access. Requirements: Expertise in ...

...G. Delete DID (When DID is going to be deleted then we will delete routing as well). H. Update DID. Note:- Primary key we will use customer ID in all tables. If a customer is inactive then the server will drop the call with the CDR status account inactive. It's crucial to note that all these tasks will initially be carried out on test DIDs. Once testing is successfully completed on both Freeswitch and application sides, we will proceed to migrate live DIDs from the current file to the Freeswitch database. Subsequently, live traffic will be transitioned from the old server to the new server. As a precautionary measure, the old server will be kept on standby for the initial months. This ensures that in the unlikely event of any issues on the new server, we have the o...

...Delete DID (When DID is going to be deleted then we will delete routing as well). H. Update DID. Note:- Primary key we will use customer ID in all tables. If a customer is inactive then the server will drop the call with the CDR status account inactive. It's crucial to note that all these tasks will initially be carried out on test DIDs. Once testing is successfully completed on both Freeswitch and application sides, we will proceed to migrate live DIDs from the current file to the Freeswitch database. Subsequently, live traffic will be transitioned from the old server to the new server. As a precautionary measure, the old server will be kept on standby for the initial months. This ensures that in the unlikely event of any issues on the new server, we have t...

want for iOS and android voip app that supports vpn , chat , dids purchase , payment gateways for iOS and android

We are looking for an experienced freelancer to create a WHMCS provisioning module for PortaBilling system. What the module must done: 1. Create/suspend/unsuspend/terminate account in portabilling. 2. Create/suspend/unsuspend/terminate SIP-account in portabilling and link with account 3. Customer must be able to choose CallerID in WHMCS. (Customer must be able to select from own DIDs or from no CallerID) 4. Customer must be able to get live CDR's (calling records) in WHMCS 5. Customer must be able to forward all calls in WHMCS (we know the settings in Porta) 6. Every first of the month, the total sum of the call costs must be imported into WHMCS for billing. Pls read api documentation before contacting:

I want a PBX system configured with SIP trunks on my server with features like Inbound and Outbound Caller Music, Hold music, Call transfer, Multiple Concurrent calls and DTMF etc. With the ability to add/release DiD's from any provider. As our goal is to manage 300+ DiDs their calls and our own portal to manage these numbers and adding funds to them. Our Usecase:- We’re a freelancing agency providing a variety of services and software solutions to clients worldwide. In order to effectively communicate with clients and provide them with top-notch customer support, the agency needs to have a large pool of inbound and outbound call center numbers in multiple countries. This will allow clients to reach the agency easily and without incurring high international call charges...

...that user) 2. Caller ID For regulatory reasons, we can't modify the caller ID to be that of another local provider.. -when users call from the App, it need to show caller ID unavailable (until that person pays for a DID, or a bundle that includes a DID.) -Then when the caller calls out, the caller ID should be the DID that they are paying for -user has ability to choose CID based on the DIDs that they own. -user has no DID, caller ID will show has unavailable ...

...that user) 2. Caller ID For regulatory reasons, we can't modify the caller ID to be that of another local provider.. -when users call from the App, it need to show caller ID unavailable (until that person pays for a DID, or a bundle that includes a DID.) -Then when the caller calls out, the caller ID should be the DID that they are paying for -user has ability to choose CID based on the DIDs that they own. -user has no DID, caller ID will show has unavailable ...

This is for a min-project that is part of a project funded by my university. I am looking for someone who can help in building an MVP as part of this project....looking for someone who can help in building an MVP as part of this project. The candidate can start immediately. The project duration is 5 months, you can work as part time or full time depending on your availability. The project revolves around creating a governance policy for a crypto wallet. The wallet creator should be able to specify the list of eligible signers (approvers) based on their DIDs. That will be the first task. Please let me know your thoughts and if you have any questions. We can have a quick chat to see if we can be a good fit. Many thanks. PS. This is for UK-based developers only (Restriction from the...

Hello, I am considering to adopt new deployment using ASTPP for billing to be integrated with FusionPBX utilizing the multi tenant IPPBX typically ASTPP will handle the billing aspects for voip termination and DIDs origination (we currently use IDT Express and ) then map it conveniently to the PBX tenant I will surely avail both cloud VMs with the base OS as per you recommendations most likely will be Debian as per the documentation or of you want me to setup boh ASTPP and FusionPBX beforehand that would be ok too and then you will just help me onboarding with the initial setup, configuration, interconnection and integration between both VMs and between ASTPP with both and IDT to be acquainted with the full picture and end to end cycle. please note that I already

i have one did setup and working with my webphone on vicidial outbound dialer - i will also use twilio i need someone who can set up multiple DIDs for my campaign along with their routing (maybe about 5 more DIDs) - as well as this same person creating me a video on how they did the DID setup and routing for me must know how to work vicidial and twilio this is for OUTBOUND dialing use the phrase "setup master" so i know you read and understand what i need please give ma an accurate price and time frame on this project add the price and timeframe to you bid request that assures me you read and understand and can do the job lets not waste each others time if you have no idea how to work either do not bid i have different groups in my dialer so i will need numbers at...

Dear all, I am seeking ASTPP expert to help configuiring YeaStar PRI Gateway TE100 with ASTPP VPS where he can demonstrate successfull interconnection and billing for both outbound and inbound calls and DIDs

...enhancements 3. Claims and Identity open standards implementation 4. Technical and non-technical publications 5ire explorations : 1. Consensus engine / Sustainability-driven Consensus Reward distribution mechanisms. 2. SDG-related Multi-agent decision-making mechanisms such as committee elections, e-voting, e-auctions, and general on-chain governance. 3. Blockchain runtime functionalities (SSI DIDs, Decentralized and permissionless naming protocols, Lightweight Cryptography, Layer 2 solutions, Protocol-level Oracles, TEEs, AI-based SGD Analysis and Telemetry with IoT & Sensory Technologies). 4. IPFS-based solutions. IPFS allows for data to be stored across a distributed network of nodes, which has the potential to eliminate any single point of failure. **Stack** ...

we are in need of a seasoned developer with solid experience and confidence working with BTC blockchain for a major project that is being backed by some of the largest names in crypto and blockchain. Need to have lots of experience with BTC, and knowledge about DIDs, IPFS, and Node.js are a plus. The scope of work will include working with our blockchain lead and the rest of our team - we have a roadmap, but what we need more than anything is someone to support where needed on a day to day basis for the next month. We're a cool, small team and would LOVE to turn this into a long-term project with the right person.

we are in need of a seasoned developer with solid experience and confidence working with BTC blockchain for a major project that is being backed by some of the largest names in crypto and blockchain. Need to have lots of experience with BTC, and knowledge about DIDs, IPFS, and Node.js are a plus. The scope of work will include working with our blockchain lead and the rest of our team - we have a roadmap, but what we need more than anything is someone to support where needed on a day to day basis for the next month. We're a cool, small team and would LOVE to turn this into a long-term project with the right person.

Need to configure Fusion PBX which will be used with Anveo Direct. SMS must be configured and working. Need an additional web admin panel for team members, as an interface to send/receive sms, originate DIDs, ability to forward to and call from a software phone. Everything via Anveo API. Initial setup and config of the trunking and SMS is the priority. All the other things are optional and to be discussed. If you are only bidding to setup and configure PBX with sms working, please mention in your bid, so that a fair comparison can be made.

We are looking for a Blockchain Engineer with a passion working on cutting edge technologies and solving hard problems. The position entails working on an agile development team within our Core Applications team to develop Blockchain applications and technologies including Distributed Applications (DApps), Blockchain tokens, and Verifiable Credentials & Decentralized Identifiers (DIDs). This role will also entail communication with executives, stakeholders and various development and business teams. 1+ years of experience with Blockchains/Distributed App (DApp) development -Javascript/Node.js for contract development for UI development -AWS competency for API development, 3rd-party integration and cloud-deployment -Database (No-SQL and/or Relational) -Experience with multiple p...

We would like to build a full e-commerce signup process integrated with our PortaSwitch () using the Portaone API () . - Client must be able t...our PortaSwitch () using the Portaone API () . - Client must be able to buy the service online and get ready to use in minutes. - All the switch configuration must be done using the portaone API. - Customer receives all communication by email for setup instructions - Admin backend to allow for customisation of plans available for purchase, DIDs available for purchase, phones - Links to our website woocommerce store at to allow users to buy IP phones and then have the phone provisioned with their account info on signup.

Hello, looking for professional who knows very well Cisco voice gateways: Cisco 2911 & 4331 + CCUM 10.5 and up for upcoming small project of porting 20 DIDs from one provider to another provider. Changes needs to be done on voice gateway and CCUM as well. For more info message me. Thank you,

i need someone to teach me how to use the magnusbilling 1. the DIDs all menu train. 2. set Plans, Tariffs, the Prefixes, if it's short number like 1250, can not call out. 3. set sound record, and now some line didn't have the sound record issues 4. the Routes all submenus 5. Voice Broadcasting 6. time in the system not right. 7. if we want to made a call, create sip account, the tariffs, the trunk, the prefix, etc, what the steps, which one is first, which one is last, order by order? 8. and also every time my local ip changes, it can not ping, i have to add IP address to the server, if not do that way, the sip account sometimes can not connect to the server, have other way to solve that. 9. sip account for spy call. 10. how to configure call out with the GOIP gat...

One of the DIDs on my FreePBX routes to an IVR. I had several users complain that when calling that number they heard a carrier rejection message and then the call hung up. When I called the number myself it worked fine, when the users called again it worked fine, now the oddest part is that all of the calls show in the CDR including the ones that didn’t work! Unfortunately I have no way to replicate this issue and looking for someone who can find out what happened by looking at the logs.

Our company is looking to have VOIP PBX server (FreePBX) configure on the Microsoft Azure. We have Around 15 extensions and three locations and I want these extension to be configured on the Polycom phones. You also need to setup Trunk to receives and makes calls. We have total 5 DIDs numbers which needs to be ported to this new server, Could someone get in touch with us for this project?

...calls and handle a few basic operations: 1. Voice greeting, coming from mp3 or other audio file. 2. Connectivity to Google Voice Recognition, listening for next commands 3. Logging the recognized voice in text and in audio (mp3 or other audio file) 4. Responding with another greeting from mp3 files, depending on recognized voice (3 options: option A option B or unrecognized) Also, we own 2 VOIP DIDs in , which has SIP connectivity and DID POP. We would like to connect those, so that calls to these VOIP will reach the Asterisk service. Other requirements: 1. The outcome for the project should be a combination of code and documentation (text files) so that an experienced developer (like myself) can follow these directions and create the service from scratch. 2. All code and docum...

...have past experience in it with installation & Repair of this & who can do front & back end both. Need some customization too, Main requirements, 1. Kazoo does not have tariff need tariff in this with all type of customers. 2. Kazoo does not have Device provisioning in it for SIP devices as per SIP brands to program certain keys. Modules Required, 1. Retail Client Client for selling DIDs & Termination routing like VOIP Switch,asiign as many extension numbers to one extension,ver cool feature. to select Different trunks for different PBXs to select number of registrations per account. option must be in Retail & PBX. user interface for PBX admin & Client & Retail client. 8. Live view of SIP call with information of error. 9. SMS service like

We are looking for someone with a big knowledge of freePBX. We have around 200 DIDs setup, and now we need to setup a queue for them. We want a structure where e.g. DID +4588327100 is setup with 3 queues 1. for all weekdays Monday-Friday 8:00-22:00 with queue number 271001 2. for weekends Saturday and Sunday 9:00-18:00 with queue number 271002 3. Voicemail with queue number 271003 Identical for all of the queues are that we use the last 5 digits of the DID for the queue number followed by either 1, 2 or 3 depending of it is weekday, weekend or voicemail.

Queremos integrar whatsappp a nuestro CRM tanto para mensajes entrantes como salientes. Adjunto requerimientos: 1-Módulo de configuración de sesiones 2-Modulo supervisión de mensajes Whatsapp 3-Poder configurar más de 1 DID 4-Asociar gestores a DIDS de whatsapp 5-Crear colas de mensajes entrantes de whatsapp 6-Que los gestores puedan iniciar conversaciones 7-Que los gestores puedan enviar documentos e imágenes 8-Si un cliente escribe que el mensaje le entre al gestor que inicio la conversación con el cliente (Poder configurar tiempo) 9-Que las conversaciones se guarden en el CRM con la información de fecha, hora, agente, caso 10-Crear variables con la información del cliente y caso 11-Poder crear plantillas predeterminadas pa...

I currently have ISDN PRI setup on voice gateway cisco 2951 and CCUM 10.5 with about 50 DIDs (numbers) I have new setup: ITSP SIP trunk to claritytel on my voice gateway and another trunk to CCUM. This is just additional new config to ISDN PRI - existing one. I have user and password from Claritytel. Besides creating both sip trunks we need to create other config so for example using digit 7 in front of any number the SIP trunk is dialed (not existing ISDN PRI trunk) I can provide more information as needed as well as detailed plan. This needs to be completed so I can use that SIP trunk from any Cisco phone in the network.

Install Asterisk in the same 3cx Server make a General SIP trunk with DID number from one of your internal extensions, then add to this trunk DIDs for all of your internal extensions; You need to add inbound rule for each of the trunk DID and configure it to route calls to corresponding extension number. Integrate on Vtiger and zoho CRM , noting that Vtiger is on cloud server and thus the Vtiger and Asterisk Server are different location, then MySql need to be open to connect Vtiger and Asterisk.

I want a FreeSwitch expert configure a Server to send calls to my ISDN gateware from FreeSwitch and receive calls from FreeSwitch to ISDN. I have 30 channels with 60 DIDs, I want Calls to be sent to the 30 Channels using the 60 DIDs randomly with 10 mins pause after each call.

...net 3. Adding New menus and new contents will provide by order giver 4. All APIs integration to our billing server a) Customer creating user and password b) Customer can buy Balance c) Customer can see calling CDR d) Calling Rate e) Reseller login option. all APIs will provide order giver 5. DID Number APIs integration. Customer can subscribe DIDs from our site. 6. Customer can buy our phones and phone lines via our site. 7. Customer can download our app via website. We will provide both links. 8. Payment Getaway Pay zone and PayPal integration 9. Provide SSL to the site 10. Social Media Links 11. Pilot plugin must add 12. MailChimp plugin 13. SEO integration 14. Customer support c...

I need someone to add Javascript web dialer using webRTC to my softswitch. This will work with Resellers, users, and admin portal. The web dialer must make calls and receive calls when DIDs are pointed to the account using the customer's extension and telephone number accordingly Please check the possibility of the dialer supporting conference calls, call hold, transfer, record, extension dialing, mute, Do not disturb and other features u deem necessary. This is a small project and estimated around $150

I need someone to add Javascript web dialer using webRTC to my softswitch. This will work with Resellers, users, and admin portal. The web dialer must make calls and receive calls when DIDs are pointed to the account using the customer's extension and telephone number accordingly Please check the possibility of the dialer supporting conference calls, call hold, transfer, record, extension dialing, mute, Do not disturb and other features u deem necessary. This is a small project and estimated around $150

In freePBX, I need the following setup: a.) I will supply SIP trunk with 2 DIDs. b.) Create a queue, and 10 extensions that will be routed with the agents evenly who ever has less work. c.) Record all phone calls, and have the ability to pull up reports by extension and FOP2 panel. d.) Voice mail needs to be sent to email, indicating the queue the DID that had the incoming call. e.) When ringing the agents it should show the DID name (brand). Please create a detailed document step by step to recreate the setup from 0. Budget is $35 to $50

Trunk is getting needs to be configured to terminate on Fixed DIDs

...full-time Rust opportunity. You will contribute to: * Design and architecture definition * Crypto- and Tokenomics solution design and implementation * Claims and Identity open standards advancement * Technical and non-technical publications Our Stack: * Rust, Wasm * Polkadot Substrate, Ink! * OpenBadges, Blockcerts, Verifiable Credentials Data Model, Decentralized Identifiers (DIDs) * JSON, JSON-LD * RDMS and NOSQL * Python, React, ES6 * AWS (and hopefully DFinity soon) You: * Live, breath, and dream decentralized * Able to work through, and implement from, white papers * Creative self-starter and continuous learner * Able to work remote first Requirements: * Experienced Rust (2018 Edition) and wasm developer * Strong data structure...

I have a couple of DIDs from DIDforSale integrated on my VICIDial. It was working fine, but all of a sudden stopped working. Looking for someone to help me with this.

New enhancements on asterisk dialplan I have 2 termination partner. Company A and B. They have their own DIDs to mask. They will provide their DIDs and we will store them in dids table to mask original calling ID. Call will come asterisk will choose one DID from did table and send the call to termination partner. it can be "company a" or "company b" but if it chooses company a DID it should send it to company a trunk. If a previous call is forward to certain DID then repeated calls will use DIDs from same company (instead of remaining limited to only that DID)

We would like training on implementing feeding in a voip trunk (from one or multiple providers) and feeding out a PRI circuit and analogue lines. We want to configure caller ID for outgoing calls and DIDs for incoming calls. Using the GUI aswell as the command line.

NOTICE that you with you bid must deliver a detailed description of how you will solve the task and which software it includes, and why this solution will work (preferably with a link, that supports this claim). Today we have one active PBX server, which receives around 500-1000 calls per day through 200 DIDs. As we within the next few months expect to upscale our business, so we daily will receive up to 5000 calls, then we want a more stable setup. We want to have our servers distributed on two locations, which are Germany and France, the servers both have a public IP. When both servers are active, then the idea is that they must divide the calls between them, so that we avoid overloading the servers, if one goes down, then the other must take over 100% automatically until the s...

Hi Ambiorix R., I noticed your profile. I have a project to develop an asterisk dialplan. Native asterisk, not freepbx etc. Phase 1 is to write the dialplan and get working. Dialplan not that complicated. Macro to route inbound calls to dids in gateway. Phase 2 of the project will be to create the dialplan/ files using php from csv extracts from database. Would want you to develop and test the dialplan in your own development environment. is this something that you would be able to help with?

Hi Mohammad Abu S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. Rate 25 eur/hr. I am having problems with new French DIDs that are set up in my ip pbx. Should be an easy fix for a asterisk expert. Thanks.

Hi Ibrahim Ali M A., I noticed your profile and would like to offer to propose to resolve an issue. I get a 404 not found for 3 new french DIDs when I try to call them. Those DID were added tp IP PBX . The belgium DIDs work fine and the settings are the same. So it seems that pbx rejecting frnech DID for unknown reason.

i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for us blames this on network. he made us move the asterisk server from behind router to public. He still continues to blame it on network.

The system is a very simple a...voicemail. All voicemail collected must be texted or emailed (with link to voicemail recording). Any code that cannot be hosted by SignalWire can be written in NodeJS and hosted on AWS Lambda. This should be very easy for someone who knows what they're doing. I can provide the full flow chart of the existing system after project is accepted, but it really is just forwarding inbound calls to other DIDs based on first answer (after pressing 1). Existing Twiml code from OpenVBX can be provided, but that does not include the menu logic. We'd like to have this complete within the next 7-10 days. You'll need to develop with a trial SignalWire account and your own AWS account, then just deliver source code and instructions. Thank you for ...

We have made lots of changes to this softswitch and wants to integrate both so that calls from FusionPBX can go through ASTPP billing and ASTPP will bill the calls. You must be knowledgeable in Freeswitch as the system is a combination of two popular programs merged together. Start allowing registration of devices, providers, authentications etc – this means one can configure DIDs, map them to a provider, run calls origination and termination, register IP phones etc. We only want developers who are knowledgeable in FusionPBX and ASTBB and have worked on these systems before with results. We will use these systems as a yardstick to measuring how good the best candidate will be. This is the interface as we have merged both at the moment but needs the integration for call ...

I need a programmer that is familiar with webooks and the LaML language. I need DIDs programmed to ring a SIP URI and or forward to a PSTN line and SMS/Text redirected to another number. All programming needs to be done on

Sila Dafter atau Log masuk untuk melihat butiran.