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    15,858 asterisk pbx δουλειές που βρέθηκαν, με τιμές EUR
    SIP with any RTP 6 μέρες left
    ΕΠΙΒΕΒΑΙΩΜΕΝΟ

    i have asterisk sip server and its working fine with any RTP i set on my system interface. but in same VM if i copy and make a new server when i try to set any new RTP its not working i dont know whats the issue audio is gone. so i need you to any how create a system where i can set any rtp as i want. like or or anything as i want. (we dont use our public IP as RTP we use any IP on our RTP like a eth0:1 interface ip is:1.1.1.1 so i will use this as a RTP) you can use any sip server or anything as you want. i just want to use RTP thats it. you can to setup this your local system or if you want i can give you server dont ask me any payment before test. if you can show me its working and audio is fine you will get payment with bonus.

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    Hello, we have a project need deploy Asterisk 1.8 from source code with support ODBC for user managment MySQL, and CDR record storing in MySQL, need support call recording in MP3 format and storing in with CDR details also need support video calls. * We don't allow in this project used software like issabel. * We need little bit documentate configuration of server in small text file with procedure of deployment * We need test all with LinPhone SIP Client . Server based on Linux OS Ubuntu

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    . A pre-installed and fully operational PBX server inside a docker container, ready to be deployed to a virtual machine. 2. The PBX installation must additionally contain: a) browser call (webrtc) with API callback to update the status and information about the call on a third-party resource b) recording API calls with a callback to update the RECORDING URL on a third-party resource c) connect multiple phone numbers and users 3. The PBX must be configured on the SIP provider XXXXXXXX page with a functional browser call (webrtc) plugin written in JQuery, which is able to accept a call, initiate a call, put a call on hold, continue the call, reconnect to the call for a specific user 5. API to initiate browser call (webrtc) via PHP/NODEJS without having to enter a...

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    Kazoo cloud pbx 4 μέρες left

    Need experts on freeswitch , kazoo pbx, etc

    €296 (Avg Bid)
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    Each file is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Vietnamese, you must also indicate the dialect) accor...rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Vietnamese, then it is necessary to indicate “Vietnamese (dialect name)”; c) if in addition to Romanian there 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: “Vietnamese (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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    Users are not able to make calls to some countries currently. The calls are not even getting to our switch (provider for international calls). We need to check what is wrong e.g if it is a setting or something else that needs to be updated.

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    Kamailio with CPS 1 μέρα left

    Currently, Customer has a few dialers using asterisk and sending call to a single provider. Combined call can go can go beyond 40 calls per second, but provider only provide 30 CPS (Calls per seconds) We currently have 2 providers (A) and (B) We need Kamailio to first group the 30 calls and send them to Provider (A) every second, and overflow / remaining within the same second to send to Provider (B) Priority will go to (A) and than (B) callflow; 1. collect and gather all the calls from multiple dialer. 2. to control and send calls in batches, 30 cps. (user define number) 3. if call exceed 30 cps within the same second, kamailio will re-route next 30 calls to another provider (B) 4. able to do media bypass Thanks for reading

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    Dear Candidates, Nautilus is cloud telephone company base in Singapore, and we provide telephone lines, extension and telephone call services to our customer over the cloud. With our current PBX system and billing system in place, we are looking for developer to help us build a self service portal that allow visitors on the website to sign up. Upon signing up they can use our billing system and PBX system in the portal. The challenge would be developing the right API to connect that PBX portion and the billing portion to this self service portal. There are a total of 5 possible modules to which you may put your consideration in the building process accordingly. 1) On-boarding also known as sign up process flow. - This module consist of an interactive design cons...

    €16 / hr (Avg Bid)
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    Make Asterisk Dialplan 4 ώρες left
    ΕΠΙΒΕΒΑΙΩΜΕΝΟ

    need to make dialplan for realtime queue using mysql for asterisk system

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    Better AMD for Asterisk 3 μέρες left

    Hi Arthur N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    FusionPBX API Λήγει left

    Hi, We have an installation of FusionPBX. We have some basic requirements to assist us to integrate into our CRM. We do not need an integration developed, we simply need examples of the best practices to make the API calls we need. Firstly a prerequisite is that all calls must be recorded. Extensions are set to ...practices to make the API calls we need. Firstly a prerequisite is that all calls must be recorded. Extensions are set to record all. 1. API call to originate an call from an internal extension to an external phone number 2 We must be able to dynamically set the external callerid through the API call 3. API call to hangup all active lines on an extension 4. API call to view all active calls on the PBX 5. API can create domain, extension, routes etc. 6. API can read ca...

    €261 (Avg Bid)
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    PBX server development Έχει λήξει left

    . Предустановленный и полностью работающий сервер PBX внутри контейнера docker, готовый к развертыванию на виртуальной машине. 2. Установка PBX должна содержать дополнительно: a) browser call (webrtc) с API коллбеком для обновления статуса и информации о звонке на стороннем ресурсе б) запись звонков API коллбеком для обновления URL записи на стороннем ресурсе в) подключение нескольких телефонных номеров и пользователей 3. PBX должен быть настроен на SIP провайдера ХХХХХХХ страница с функциональным плагином browser call (webrtc), написанным на Jquery, который способен принимать вызов, инициировать вызов, ставить вызов на удержание, продолжать вызов, повторно подключаться к вызову для определенного пользователя 5. API для инициирования browser call (webrtc) через...

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    I need a asterisk developer Έχει λήξει left

    Fusionpbx integrations and customisations

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    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed

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    There are 163 entries in total. Each duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Italian, you mus...following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Italian, then it is necessary to indicate “Italian (dialect name)”; c) if in addition to Romanian there 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: “Italian (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...

    €146 (Avg Bid)
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    26 προσφορές
    PBX yeastar s20 and mikrotek Έχει λήξει left

    PBX yeastar s20 and mikrotek problem we need to connect both yeastar s20 with mikrotek between two countries and i think there is a problem with the ports and vpn so we can't use linkus between the both

    €53 (Avg Bid)
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    ...build a proof of concept Java multithreaded standalone application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this Java application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. When an event is received from the asterisk server, the application will send the same event to all or some of the clients connected. When a command is received from any of the clients connected, the same command will be sent to the asterisk server. In other words, I need a "wrapper". This is just a p...

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    ...Laravel + NodeJS/VueJS (Recommended) Data to be provided: Chemical Properties Profile Database (.XLSX) Chemical Supplier Profiles (.CSV) Import/Export Intelligence (.XLSX) * Initial Figma UI drafts and template to be provided. * Must be comfortable with signing an NDA. * Must include PerfexCRM options and data intelligence options to be parsed from data silos. * Ability to integrate with VOIP/PBX. SaaS platform should be able to integrate with data intelligence sources to be provided in .xlsx and .csv formats and ideally big data formats. Preferable Qualities: Understanding of Relational Databases. Understanding of Docker. Understanding of Omnichannel. Potential freelancer must be good at both front and back-end development. Attention to UI and operability must also be ...

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    I need to setup Nagios or similar monitoring software to monitor my network. The Nagios server will be hosted in AWS and I will need to monitor systems (Windows & Linux) on the internet and from behind NAT. -servers -apps (Asterisk, Apache, MySQL etc) -Network outages I need someone who is familiar with this type of work and will provide me with better ways to accomplish this is possible.

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    Hi guys We are looking to build zoho and zendesk integration with our PBX Support API there are some Zoho , zendesk and PBX integration features: 1. Auto create ticket when phone call coming 2. Auto pop up the ticket page when this phone caller already have a ticket 3. Auto create a contact person when the phone caller number is not in crm database.

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    ...Add outside 3rd party SIP VoIP service to our desktop client Electron JS Soft-Phone App for Mac and PC versions to download from our website. We currently have a VoIP SIP Soft-Phone that works with our PBX, it also does API calls where we data mine the phone number and show information about the caller from that phone number, call logs, SMS / MMS feature, Auto Dialer and more. What I’m trying to do is make it so we can put our soft-phone on the Mac App Store or provide a Mac & PC download so people can pay us a small monthly fee for the Soft-Phone but use their SIP credentials from their PBX or VoIP service while keeping all our features working. Or have it like a LIN Phone where anyone can pay a fee then enter their own SIP VoIP service provider credentials...

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    Build an IVR to receive calls Έχει λήξει left

    Build an IVR and chose a server. Add recordings and see live stats. Would prefer to see an example. Ongoing work available in this startup with experienced management etc is required for this project that is bound for success. THIS IS TO RECEIVE CALLS, NOT TO MAKE CALLS. CLIENTS CAN BE ADDED AND THEY CAN SEE LIVE AND HISTORICAL CALLS. Please read before bidding, this is not a generic PBX project.

    €1443 - €2886
    Προεξέχον Επείγον Σφραγισμένο
    €1443 - €2886
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    Call forward via API on asterisk Έχει λήξει left

    Hi Arshad N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    ONLY for skilled DEV in Asterisk n Freepbx ######## Help in FreePBX/Asterisk CDR REPORTS Please check ScreenShots (ivr, logs, cdr) ######## Im realy BLIND in CDR reports about: where pbx is calling to (what number), where caller is going to (what IVR/Destination 1 or 2) All outgoing calls from Dongle1 are pointed to IVR, when someone pickup call it goes to IVR ("Message: Please choose 1 or 2"). ######## TASKS: A. DID number on CDR report - need to know where (what number) pbx is calling to B. DESTINATION - need to know what button/destination/extension caller goes (#1 or #2) this is tricky couse AT THE MOMENT i can see in DESTINATION number 9 (when user press1) and number 10 when user press2 C. LOG UNANSWERED CALLS in CDR report ######## IMP...

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    ERPNext VOIP Integration Έχει λήξει left

    Wish to have an integration of an Asterisk PBX like FreePBX or something with ERP Next. We should have an incoming call popup with data being picked up from the ERP Next CRM Module for an existing number and We should have an option to save an unknown new number as a new Lead or Save to an existing customer/supplier / another contact similarly, we should be able to do a click-to dial from within ERP Next

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    VoIP Softswitch DID billing Έχει λήξει left

    We are currently looking for a new Softswitch, especially for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating ...Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating one from scratch. We prefer if the project used Freeswitch / Kamailio instead of an ast...

    €2397 (Avg Bid)
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    Backend Python Developer Έχει λήξει left

    We are looking for an experienced python developer that will be responsible for creating a Python basked Backend API for a Telecommunications platfo...are looking for an experienced python developer that will be responsible for creating a Python basked Backend API for a Telecommunications platform. You can think of it as Twilio + Vonage. This platform will be used to provide telecom functionality for our customers. We use MySQL (Tortoise ORM) as our backend database, Asterisk for our Media and RTP traffic, Kamailio for our front end Registration Servers. You will also be responsible for creating a client side API to access PBX features. This API will be used by our front end interface for customers and also will have Admin access for our internal CRM system to provision and ...

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    FreePBX Admin -- 2 Έχει λήξει left

    Looking for someone available during USA business hours. We have a PBX that needs to be updated and worked on.

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    Jak w tytule, potrzebuję zmodyfikować kod źreódłowy softu Asterisk (C / C++) żeby: - wysyłał dźwięk do zewnętrznego API (Google Api Speech-to-text) - na bieząco odczytywał rezultat z tego API i umieszczał wynik w zmiennej Asteriska - przykłady bibliotek Google API są umieszczone na Google Cloud Docs

    €529 (Avg Bid)
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    I would need to Integrate Vtiger CRM 7.X and Issabel PBX. functions NEEDED on the Vtiger interface: - popup with caller information (only for the agent who answers) - click2call - History (including recording) What will I provide: - a fully working VPS - vtiger and Issabel (Asterisk) already installed on the VPS Important additiona informatins: - All the extensions MUST be PJSIP - I have a working version of the integration between vtiger and Issabel, but it doesn't work with PJSIP extensions. In case you are interested into working on this version, just let me know! - If you plan to customize the Asterisk callflow, you have to warn me PS: For those who don't know issabel, it is a Freepbx fork. So it works in the same way

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    issabel PBX webapp Έχει λήξει left

    we want to build a module or something to popup the caller information or add the caller information if it doesn't exist.

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    Hello Ervin, I would like to ask you if you may be interested in a project like this: I would need to Integrate Vtiger CRM 7.X and Issabel PBX. With the following functions: - popup on vtiger dashboard when the agent answer - click2call - History (including recording) Vtiger and Issabel are already installed and configured by us on the same VPS. Only integration would be needed Thanks in advance Andrea

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    FreeSwitch Monitoring tools Έχει λήξει left

    I need tools to monitor FreeSwitch PBX

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    This project is about "ISP CRM" With freeradius, Asterisk and Network Monitoring Capability which Support Vendors [Mikrotik, Huawei, Cisco, Juniper and Star OS]. Required Features and Modules # Admin Control Panel (Web + Android ) * User Management * Plan/Tariff Management * Finance Management >Invoice generation and payment processing >Recurring billing engine >Prepaid billing engine >Data Cap and Top UPs >Voice billing (VoIP CDR auto processing) >Payment gateways and credit card processing (India). >Cash Desk module * Accounting software integration (QuickBooks and Tally) * Referral Management * Sales Management and Onboard Process * Support Management * Task Management * ...

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    Vicidial (Asterisk) Customization Έχει λήξει left

    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed Before hiring you to our long term projects, there will be a small test which needs to be cleared. Please share screenshots, once you are successful with the test. Test Use REST API, collect all Leads with below fields, from and add display it in your Django app. Username - Demo Password - Demo (For MD5 password, you can translate ‘Demo’ online) REST URL endpoint: Module name for connecting to SuiteCRM Leads Field names to collect are ‘ phone_work, first_name,

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    asterisk dial plan Έχει λήξει left

    i have a simple asterisk set up where users can connect with sip phone and make a call, i want to temporarily change the dial plan so that users can just hear a audio that i saved in sounds when they try and make a call

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    asterisk calling issue Έχει λήξει left

    Require help regarding calling

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    Survey with Asterisk Έχει λήξει left

    Hi Abu K., I need some help to configure a phone survey on a client's Asterisk server and integrate the results with Stats () Can you help me with that?

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    Setup a Trunk FREEPBX to Flowroute Έχει λήξει left

    I have a Dell Sonicwall FreePBX PBX and Flowroute Trunk. I need help configuring the Trunk / Firewall to work as intended. Only Reply to this Job is you are able to start working immediately.

    €119 (Avg Bid)
    Επείγον
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    Installation and maintenance of an Asterisk telephone system. Takeover of the old Asterisk system. (The old system is 15 years old). Many dial plans need to be rewritten. New functions are to be added, e.g. chat rooms for conferences etc. Provision of the data e.g. on an ASPX interface for other systems. We would like a contact in German. --- Installation und Pflege eines Asterisk Telefonsystemtems. Übernahme des alten Asterisk Systems. (Das alte System ist 15 Jahre alt). Es müssen viele Wählpläne neu geschrieben werden. Es sollen neuen Funktionen hinzukommen z.B. Chaträume für Konfenzenzen etc.. Neue SIP Telefone installieren. Intergration eines SIP Anbieters wie SipGate sowie weitere Verwendung des ISDN Adapter. Bereitstellung...

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    Asterisk with Google Speech to Text Έχει λήξει left

    I need someone who can sucessfully setup Asterisk and Google Speech to Text. More addons will be done later which is payable extra.

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    Install TLS 'let's encrypt' certificate, setup PBX to use TLS with SRTP

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    we have built new virtual machine Issabel PBX . A- The old system is Elastix , we want someone to manually move the call logs and recording from The old elastix to the new issabel B- We want someone to help us increas the virutal machine size for the Issabel ,becuase the current machine size is 100G and we want to make it 500GB or higher ,we know how to do it on vmware but internally how it will increase we don't know .

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    Looking for a skilled Voip person to configure two VOIP terminations to Open VPN on Free PBX Terminations are Snom and Yealink. VPN and system are already working with Fanvil terminations, I wish to configure on Snom and Yealink terminations.

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    Hello Friends, I am looking for a full time VoIP telephony Developer who can create a web based SAAS Application for Call Management Portal that can help business to manage their huge call volumes. For more details, I can also show you the reference applications.

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    I need someone to configure Yeastar P Series Cloud PBX with my gateway.

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