Asterisk web sippekerjaan

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    20,690 asterisk web sip tugasan ditemui, harga dalam USD

    Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

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    Diseño y desarrollo de una aplicación Aplicación móvil OTT Windows – Linux – Android (todas la posibles versiones) -IOS, iphone, - IPN(las diferentes versiones) con las siguientes funcionalidades: - Cliente SIP para VoIP, con la opción de elegir entre servidores SIP - Cliente de chat (REST API), el chat debe ser capaz de enviar y recibir Audios, texto, emojis, imágenes, vídeos, ubicación y archivos - Soporte de videoconferencia - Notificaciones PUSH - Soporte de cliente VPN

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    Hi I have an ongoing problem from few hours. My server appears to be performing a portscan toward 'random' ip addresses and udp ports. The process that is sending those packets is Asterisk which was already at version 16.29 and I've upgraded it to the last version, 16.30 but the problem persists. I can't provide ssh access to the server but I can provide whatever packets captures you prefer. I'd like to know the source of the problem, if it's some virus in the office of the client or some external attack. Max 100 euros.

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    Hello I will need a mobile app for android to transfer calls from asterisk to skype (viber) and from skyp(viber) to asterisk...alone audio calls. I will wait your answer.

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    Hi, I have several asterisk 11 servers. I need: 1. Export inbound and trunks 2. Merge duplicate data. 3. Import data to asterisk 16/20 server. Notes: Code will be reviewed and approved. Code will be tested on tests servers. Developer will not have access to production servers. DM me for any further questions) Thank you.

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    6 bida

    Looking for a sip softphone for windows desktop with g729 codec ,call transfer and conference option. Should work behind nat. Stun and rtp ports should be auto filled and static. We will build with basic options now and will upgrade more , so bid with your best budget.

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    Hello Freelancers, I’m looking to implant live DTMF logging into my agents panel using asterisk. The DTMF will be read from asterisk server and will be shown in my agents screen in real time.

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    We need a Linux expert to install faxing in asterisk..

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    Hello all, I would like to capture live DTMF information through asterisk and make it appear on my agent login panel. Essentially all it will do is when a call gets connect and an agent ask the customer to enter a code through their key pad it should appear on the agents panel.

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    We have an extension which is registered to the asterisk server.. but when making a call the Asterisk returns 401. The task is to find out: 1: why this is happening 2: what we need to do to fix it 3: Make test call to prove fix We will only pay for a fix..

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    I have an Alpine docker image with Hylafaxplus and iaxmodem and asterisk. my phone provider is twilio. I try to send faxes, but there are a lot of errors. here are some errors. Unspecified failure to train with receiver No response to PPS repeated 3 times. No receiver protocol (T.30 T1 timeout) I imagine that there is a problem in the configuration, can you help me? thank you

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    What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time ...

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    Can not send “@” in sms by Dongle Usb Example: I send: test @ I get: test ¡ What im missing in dialplan ? [textmessage] exten => 111,1,NoOp(SMS receiving dialplan invoked) exten => 111,n,NoOp(To ${MESSAGE(to)}) exten => 111,n,NoOp(From ${MESSAGE(from)}) exten => 111,n,NoOp(Body ${MESSAGE(body)}) exten => 111,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) exten => 111,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) exten => 111,n,NoOp(Send status is ${MESSAGE_SEND_STATUS}) exten => 111,n,GotoIf($[“${MESSAGE_SEND_STATUS}” != “SUCCESS”]? sendfailedmsg) exten => 111,n,Hangup()

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    I need to Configure SIP trunk between the billing platform MagnusBilling and the MultiTenant PBX. I need to set it up so that all tenant PBX's are able to call out from the PBX and get billing individually by the MagnusBilling platform.

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    Here are goals for the modification, 1. Modification for both MacOS and Windows 11 platform. 2. Provide installation packages for both MacOS (.pkg with cert) and Windows 11 (x86 - msi/exe). 3. Rebrand Blink to our product brand (2 brands). 4. Modify login to use Web Services, this web services will be provided by us. a. Web services will providing SIP credential and phonebook. b. Web services will be using encrypted messages. 5. Removing unnecessary GUI and functions. 6. Synchronize contact/phonebook with cloud Web Services. 7. Change the color scheme of the app according to our specification. 8. You will need to handover the codes. 9. You need to prove the codes are working and can be compiled using virtual machines setup by us. 10. We wi...

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    We require someone who can integrate php agi in asterisk box to replicate the similar commands to control dialplan as given under below link here, To enable our customers to run their own IVR using API and SDK Architecture using Cent OS, LAMP framework, Asterisk and PHP-AGI User Accounts connects using API and run their own IVR business logic Our VoIP Arch -> Connected to multiple sip endpoints for each users on their account. I need some one who can help me setup an asterisk on linux machine probably in some better datacenter such as aws or azure that could run behind an proxy public ip for client server connection. And than use php agi to develop an sdk that could send and receive rest api and xml commands for controlling ivr dialplans

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    Need create notification for ippbx support ios and android (React Native) want to make notification to support app wake When the screen is turned off or the screen is locked when someone calls through the sip application, both ios and android

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    Hi We are trying to find out the possibility of developing a middleware (B) for our system. We have (A) a Voip Switch (Originator) (C) A VoIP provider (Terminator) (B) will be sitting in the center and 'listens' to each Ring Back Tone (RBT) when a call is established and 'ringing'. A--B--C Originator -- Middleware -- Terminator RTB frequecy will be based on standards according to Internation Telecommunication Union (ITU). B will reject calls when RTB frequency are not met.

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    Need some assistance to getting our app to work Incoming and Outgoing calls. demo like ctxSip

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    I have developed an automated system to make calls in asterisk, this is with a , the problem is the calls do not go out and I was investigating, the reason why the calls do not go out is because they do not include exten => _91XXXXXXXXX,1, AGI(agi://). the problem is that I don't know how to include it in the

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    * Modern Looking Softphone App. Flutter language to be cross platform, SIP coding, should include: - User Registration form; - Keypad to enter phone number; - Access to phonebook; - Separate page to send and receive SMS/MMS; - Show Balance; - Recharge Balance; - Group call; - Low Battery consumption for background activity of app; - Add users through phone number; - Indicator if other user is online/offline; - Ringtone selection; - Voicemail; - Call waiting; - Users can add multiple phone numbers after purchase; *Multi-Page App includes the following pages: - Softphone (keypad page); - Users page (where users can see other online/offline added users/add users, chat through SMS/MMS, or create groups to make group calls); - Wallet page (shows balance, users...

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    We are looking for feeepbx with Gsm gateway setup for incoming and outgoing calls you should have experience on FreePBX and make FreePBX secure . And we want to use WireGuard on same server to connect out sip extensions and gateway as trunk

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    our company has an asterisk phone pbx connected to local carrier and additional voip carriers. we are migrating all our systems (accounting, sales, etc) under Odoo. We are looking for someone to configure Odoo VOIP module and connect to carriers.

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    require sip trunk call to china with randomly display CLI

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    I need someone to recommend, install and configure OpenSource asterisk based billing system.

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    Create a small demo project in Delphi 10.2 Tokyo or higher that uses the BareSIP libraries () to create a basic SIP client that registers a SIP extension to a SIP server (FreePBX or 3CX for demo purposes), makes a call to an extension, sends a WAV file as G.711u audio over the call, and hangs up the call.

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    MIKOPBX Call routing Training We are seeking a Russian or Ukrainian freelancer with specific experience in configuring and working with MikoPBX. The freelancer will provide training. Should have previous experience in Asterisk Dialplan and configure gsm gateway for incoming and outgoing calls to setup for call center this is open source PBX need to work on it

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    hi we need call routing training and one login page for extentions to see thier CDR and call recordings that already available in main Admin panel we just need to show them under each user its MIKOPBX opensource based on asterisk

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    Hi I'm not a Cloudflare expert, I have an account with spectrum enterprise enabled (trial) and an Asterisk pbx server with a websocket webphone and stun. I need help to configure Cloudflare in order to proxy the voip calls. Max 100 euros. Please bid only if you have experience with Cloudflare and custom applications (spectrum).

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    Requirement: to be a native speaker of the Turkmenlanguage. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ‚Äč‚Äčrepresented on the record (for Turkmen, you must also indicat...the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Turkmen, then it is necessary to indicate “Turkmen(dialect name)”; c) if in addition to Turkmenthere is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Turkmen(dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    Sip panel engineer, drafting. Need stamped drawings. So the plans can be submited to manufacture and city zoning committee Structurally insulated panel design Structal insulated panel engineer or architect. Familiar with SIP panels.

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    Deploy a cloud voip solution in aws configure application for multi Tennant pbx deploy sbc and configure and test

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    The goal of this project is to create a step-by-step guide for configuring Kamailio (a free and open-source SIP server) to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown as it will be used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Kamailio or Martini Security's offerings. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Kamailio with STIR/SHAKEN. Martini Security offers a certificate enrollment

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    ...using FusionPBX/FreeSwitch or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using FreeSwitch with STIR/SHAKEN (). Martini Security offers a certificate enrollment client called Olive () which can be used if FusionPBX/FreeSwitch has working STIR/SHAKEN support built-in. If it does not have native STIR/SHAKEN support, the Martini Security STIR/SHAKEN signing and verification server called Vermouth () can

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    The goal of this project is to create a step-by-step guide for configuring Asterisk/FreePBX to use STIR/SHAKEN based on Martini Security's offering. The guide should be written in markdown and used to generate a PDF similar to those found on Martini Security's website (). The guide should be clear, easy to follow, and suitable for someone with limited experience in using Asterisk/FreePBX or Martini Security's offering. To complete this project, you will be given access to a pre-production environment at Martini Security where you can obtain the necessary API keys for enrollment. You will also have access to existing documentation on using Asterisk/FreePBX with STIR/SHAKEN (https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN). Martini Security

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    ...will provide a .pcap file captured with tcpdump. The file captures VOIP calls made on our network. You will write a command-line utility program that reads the pcap file and outputs the RDP audio streams into .au files. The audio is in 8000hz PCU format generated by SIP telephone calls. For each SIP telephone call you will provide THREE audio files. One file will contain the RDP audio for each separate side of the conversation. The third audio file will contain BOTH sides of the conversation. In addition, for each SIP conversation, you will provide a data file containing JSON text, showing the TIME and DATE of the call for the PACIFIC timezone; also TO and FROM fields showing the names of the caller and person called, if these are available in the ...

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    Early-stage, Midwest, startup gearing up to produce SIP wall, floor, and roof components.

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    Jisti Setup Tamat left

    Installation eines Jitsi Meet Servers auf bereitgestelltem Ubuntu Server. Zugangsdaten stehen bereit Grundkonfiguration Jitsi für Browser und App-Nutzung Nach Rücksprache: - erstellen eines SIP Accounts - Setup FreePBX mit IVR Einwahl

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    Telecom billing and routing software ASTPP installation and setup for community and enterprise license - Full setup with sample customer. -Gateway integration -SIP proxy. -Reseller account. -Payment gateway etc.

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    We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

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    WHATSAPPBX Tamat left

    We use Asterisk PBX along with Free PBX using SIP Protocol. We would like a developer to integrate Whats App calls to our PBX System. Either directly by configuring the Gateway on the PBX itself or as a trunk which is piped from an Android Phone (in this case the phone will be ON and connected on WiFi) to receive and make calls and pipe it as a SIP trunk to our PBX.

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    I need a cloud FreePBX system to manage my sip numbers. In the meantime, I need to use database to store all of the CDR (including incoming and outcoming, etc.). Moreover, the instant incoming calls need to show to a webpage on my front-end app, and the response from the app (such as answer, hang up, hold on) will send back to the Asterisk.

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    Buscamos un perfil que cree contenido referente al contact center, debe tener conocimiento de dicho sector, customer experience, employee experience, software de contact center, trunk sip, centralitas, departamentos de contact center, etc. Que sea de idioma español nativo. Conocimientos del contact center. Queremos que este con nosotros mensualmente, hay mucho que redactar al mes, artículos, cursos, grabar podcast, pdf de contenido para su descarga. Tenemos 7 lineas de negocios y cada una de ellas es de una rama de contact center.

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    Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

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    Hi I am looking for a development specialist for Asterisk PBX (I work with vitalpbx based on Asterisk) for the purpose of 3 developments 1. Press 2 call 2. Connection to the CRM system 3. Connection to the ZOHO system Regarding section 1 Sends the system API:

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    ...receive speech packets and data. 3. We need to make certain enhancements to the interface and communication between the UniMRCP plugin and the tone / speech detection library we own. 4. This project will require you to work about 12-16 hours a week for 6 months. 5. Skills: (a) Pro in C/C++ (b) Familiar with UniMRCP plugin development architecture. (c) Telecommunications Engineering, Asterisk, Socket Programming, System Programming IVR Software, gRPC, Websockets. 6. Deliverables (a) Fully-functional plugin in executable form along with libs. (b) Product must be containerized. (c) Complete source code of interface. (d) No GPL code to be used (e) Documentation: (a) User Documentation or README file to setup/run, (b) Design and source code documentation. ##Yo...

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    We experiencing the issues with audio in IVRs and MOH under high load(300-400 concurrent calls). They are breaking up and very choppy. The CPU load with 400 calls is around 50-60%. There is no issues with about 150 calls. The server details are below: CPU: Dual Xeon Silver 4309Y (16x2.8GHz) RAM: 32 GB SSD: 1TB(RAID 1) Asterisk: 16.21.1 Front End: Issabel It should be enough resources to handle this load. We need someone who can analyze the issue and provide the solution for it.

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    We have an app dev/integration project: Interested? Introduction Adobe Connect, a web-based collaboration, and interactive meeting software, provides the facility to integrate telephone-based audio conferences with the live meeting room. You can control the audio conference using a Connect meeting room through this integration. Controlling the audio conference includes actions like starting/stopping an audio conference, recording, muting/unmuting audio conference participants etc. Getting Started Adobe Connect’s telephony-related functionalities are handled through a separate service, namely, Adobe Connect Telephony Service. Within the telephony service are the various telephony adaptors, which act as the glue code between their respective conferencing service interfaces an...

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    ...include integrations with external APIs and the ability to create and sell products through a WordPress module. In addition, the software should include stand-alone modules for live chat and SIP communication, a learning management system, and appointments. There are several additional features and functionalities that are outlined in the attached document. It is important that developers carefully read and understand the entire document before making a bid on this project. The document includes details on the Master SaaS, Invoicing, Safe Mail, Modules, Stand-Alone Modules, Templates, Accounting, Chat and SIP Client, and Onboarding Wizard, as well as using external APIs and storage. Developers should have experience in building software solutions with similar features an...

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    hello! i’m looking for a drawing outline for a canvas. this is for a paint & sip event so i just want a black and white outline. i’d like something like someone wearing jordan 1 shoes and a little bit of their legs with a sock on it with a K logo which is attached in the attachments

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