Asterisk voicexmlงาน

I need you to develop some software for me. I would like this software to be developed using Python.I need an automation for homeassistant. The automation will trigger an asterisk call when a button ftom a sonoff is pressed

Our objective is create a new user interface based on Asterisk/Freepbx. This totally new interface should have all the asterisk/Freepbx functions and possibly add other new features like videocall, webrtc interface for users and APIs to integrate it with CRMs like vtiger.

Having servers preinstalled with both Freepbx and Vtiger on Windows The needed is below : Integrating FreePBX based on Asterisk and Vtiger 7.x to help achieve the below: Click to Call from CRM • Incoming Call Popup with Contact/Lead/Accounts Details • Call pop up shows Previous Description • Call logs with all details • Create Account , Lead ,contact, Task Options in call Popup • Call Hangup and Call transfer Option in Call Popup Able to Save Note in call popup Creating users and having them access to their own record or other as per configured In a need of a connector (developed or open to be used ) and detailed steps in maintaining the connector; after installation as well as other necessary guide on the...

Require a document to serve as a how to in order to integrate Jitsi Meet / Jigasi / Asterisk / ejabberd. Asterisk version 13 / 16 Should include all aspects of configuration - assume working asterisk installation. Milestone: 100% payment on successful test implementation using the document.

For a call filter - Expert in Asterisk (Macros)

Need to configure SIP trunk between Asterisk and Seimens PBX. If any once from Saudi Arabia can do it, let me know.

We have a Android Chat app developed in Kotlin and on asterisk. We need a backend panel with following features in it... Panel Requirements 1 Profile Pic moderation- cancellation/approval provision 2 Redemption- Status change provision (Success/Failure/Processing) 3 Purchase history: Mobile number wise and date wise data transaction with Amount/Mode and status. Only View purpose 4 Notification- Push notification provision to customers 5 "Create 3 different users of Panel: 1. Admin A/c, having all right 2. Executive A/c, Having on Profile Pic moderation rights/ Notification provision rights" Apart from this, we need a guy who can also work on JAVA APIs as well. So that we can create APIs to get the data in App from database. This work need to be done within 5 days

I have a sip freepbx server and i want to convert a sip trunk to pjsip. The trunk have different username and auth name. And in this contain the @. Upon request i can provide you the full sip trunk config. The freepbx is in internal network so i can't give direct access but i can provide logs, tcpdump for wireshark etc. I use the latest freepbx version (15) with asterisk 16

We are setting Multiple asterisk system per customer and would like to move to a multi tenant system like Elastics or Kamailio. We would like to install A2 billing and segregate per customer. Each customer is associated with a different Trunk.

Having servers that can be preinstalled with both Freepbx and Vtiger on Ubuntu. The needed is below : Integrating FreePBX 14 based on Asterisk 13 and Vtiger 6.5 or 7.x to help achieve the below: Click to Call from CRM • Incoming Call Popup with Contact/Lead/Accounts Details • Call pop up shows Previous Description • Call logs with all details • Create Account , Lead ,contact, Task Options in call Popup • Call Hangup and Call transfer Option in Call Popup Able to Save Note in call popup Creating users and having them access to their own record or other as per configured In a need of a connector (developed or open to be used ) and detailed steps in maintaining the connector; after installation as well as other ...

asterisk developer needed to be able to display the dtmf live on screen on web whiles in call

Hi , I'm looking for someone to install and server Kamailio SIP Server to acted as a SBC for SIP Trunks and authenticate users that will register to the Issabel/Asterisk PBX. I also need need replication between the 2 PBX.

Addition of the following features to the existing linphone application(using asterisk): -Video and audio conferencing (meetings) with a maximum of 500 users -Screen sharing /presentation -Tools for presentation on the side bar e.g Writing, File sharing, and collaboration -Fix of bugs on contact API(when you start application). Application crashes when you have many contacts -Fix of other crash issues on current application (STRONG KNOWLEDGE OF LINPHONE AND ASTERISK IS A MUST)

Addition of the following features to the existing linphone application(using asterisk): -Video and audio conferencing (meetings) with a maximum of 500 users -Screen sharing /presentation -Tools for presentation on the side bar e.g Writing, File sharing, drawing -Fix of bugs on contact API(when you start application). Application crashes when you have many contacts -Fix of other crash issues on current application (STRONG KNOWLEDGE OF LINPHONE AND ASTERISK IS A MUST)

The ghostwriter must be an expert in Indian Political Systems with in-depth knowledge on the history of po...Tables and Figures should be centred, and numbered and titled in bold capitals. Acknowledgements should appear at the end of the main text and before the references. Any appendices should appear before the References. Footnotes should be numbered consecutively in the text and placed on a separate sheet of the manuscript. Any footnote attached to the main heading should be designated by an asterisk. References follow the author-date Harvard style. References in the text should give the author’s surname, year of publication and page number if a direct quote is included. References should be listed alphabetically after the text. Journal and book titles should be wr...

Hello I NEED CONNECTED MODEM 4G HUAWEI for Asterisk AND USE IT IN VOICE Now i use only 3g chan_dongle i need 4G

We are Cloud telephone services provider and would to get developed a VoIP softphone for Asterisk platform. It should work on iOS, Android and Windows laptops. Please contact us if interested. We can share detailed requirements with interested developers. Regards Sanjay

Hellp me to call from asterisk , ssh to international number !

Hello I NEED CONNECTED MODEM 4G HUAWEI for Asterisk AND USE IT IN VOICE Now i use only 3g chan_dongle i need 4G

We are looking for solution like a traditional gsm or CDMA voip gateway. This project will be separated in two is mobile applicatio...application will register to a server and accept call from that server with IXA OR SIP protocol. After that call terminate to GSM Network. (this part like traditional gsm gateway). This mobile application will work only wifi internet connectivity. Beacuse gsm intenret date normally disbale during any gsm call. All call will pass through gsm 729 codec. The registration server may be voip switch or asterisk or any other server. The server will receive call from another voip switch server with sip protocol. Certain number of registered mobile will be able to assign in a group of gateway. On server have have include option show balance through ussd...

We have the basic knowledge of Asterisk and can setup the PBX to make outgoing calls using the PRI Lines, click 2 call etc. Now we need to setup a portal, like Knowlarity / Myoperator etc where we will sell this as a service. a) Sell a IVR (Welcome to our company, Press 1 for sales, Press 2 for support etc...). Asterisk and web development Knowledge is a requirement for this project. b). Once a call comes in its routed as per the rules required and a outgoing call is made to the users number and both the calls are patched. meaning for every call incoming there will be 2 calls. one incoming and one outgoing to the users mobile and both are put into conference and are recorded. c) similarly we could make a outgoing call using the board number by calling the board number and th...

We have a FreePBX installation. Our customer has a CRM used by call centre consultants and hosted at a different cloud provider. All telephony based activity is done via the CRM (no physical phones or 3rd party soft phones) to connect to the FreePBX server. The customer has a implementation, using WebRTC. The implementation is making use of 2 secure connections (wss://); 1. to handle the voice and SIP 2. to handle server requests such as Login, make call etc. and receive call progress data such as channel added / removed, connected An important feature for the call centre is that all sales calls are starting as a conference, this to create a customer experience that allows for adding additional persons without the unpleasant silence etc. associated with being put into a conference room. ...

No companies, individuals only Your profile: Young, talented, efficient, you are looking for a challenge. We are a remote team working on interesting innovation projects and are passionate about our job. We enjoy what we do and we do not count our hours but we measur...: 1st interaction: "register user" Ask what language would he like to speak. In the language of choice: Ask for birthday Ask for Name Ask for his address 2nd call : Tell the user what you know about him Hey {name} you are currently {age} in the right language. what would you like to know? User can ask: What's my name? What's my age? What's my address? OPTIONAL Angular React Ionic Openvidu Asterisk Sails Web socket php symfony You are expected to work in GMT+2 timezone. What budget do ...

Hi guys. I'm searching for an experimented person on Patton SN4171 e1 gateway configuration including trunking with ISSABEL IPBX (Asterisk based) and call routing configuration at the patton side.

No companies, individuals only Your pr...successful with you and grow together! We are looking for an agile developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different techologies by self learning. We are looking for attention to detail, communication and reliability. Mandatory expertise Nodejs Angular Sails Websocket Mongo OPTIONAL React Ionic Openvidu Asterisk Chatbot php symfony You are expected to work in GMT+2 timezone. We pay 2500 CHF per month and offer 20 days of paid holidays per year. TO APPLY 1. Let us know your experience in years and short description of your work performed on all mandatory elements and ideally on optional elements. 2. Send your CV 3. Put your bid for monthly in...

hello i have a raspberry pi that have Rasterisk on it i am using chan_dongle solution to use Huawei E160 for a GSM port my end is this : Port1 is talking B, during this time a sip call(from C to D) is coming to Asterisk , Asterisk should hold call on port1 and B , and port1 should call D when call connected to D , port1 should merge all calls (Conference) it means now port , B , D , C is on one call

I am looking for someone to install software link below including with all dependencies e.g: asterisk, soundcard all installation process is explained just need someone who did this past. Please with no experencie or familier please dont bid,

We want to integrate asterisk free PBX with Avaya and siemens PBX for one of our customer... If any one local KSA available let us know..

The task is to search a VSTS repository for exact and fuzzy keywords in the source code files in the repo. For example, if searching for the keyword "ADD", the tool needs to scan all the files for "ADD" as well as "*ADD*" where the asterisk can be anything (or nothing), but not including spaces. ASCII characters only in the keyword list and the VSTS repo. Case is ignored, so “ADD” should match “add” and “aDd”. The keywords will be in an ASCII file and need to be read from top to bottom, then each file in the VSTS repo scanned for an exact match of the keyword, then for fuzzy matches. The tool can scan all exact matches first, then all fuzzy matches, or it can scan for the exact then fuzzy match one after another,...

We need to install asterisk to the debian 10 server. We need to start the asterisk on server, add 2 SIP accounts and organize SIP calls between them.

I need a team of developers with speciality at developing Asterisk based call taking applications , I also need a Computer Aided Dispatcher Application Developer with speciality to implement GIS based solutions.

... Tables and Figures should be centred, and numbered and titled in bold capitals. Acknowledgements should appear at the end of the main text and before the references. Any appendices should appear before the References. Footnotes should be numbered consecutively in the text and placed on a separate sheet of the manuscript. Any footnote attached to the main heading should be designated by an asterisk. References follow the author-date Harvard style. References in the text should give the author’s surname, year of publication and page number if a direct quote is included. References should be listed alphabetically after the text. Journal and book titles should be written in full. Some examples are: Baumol, W.J. (1986) Microtheory: Applications and Origins, Cambridge, MA: ...

Looking for Asterisk PBX Developers who can do deep customization and integration to various in-house applications.

Realizzazione di una WebApp, che permetta di comunicare con PBX Asterisk: Webphone WebRTC BLF Status

I have a working Jitsi-meet server, and a working (production) Asterisk Server. Would like to be able to dial into a converence. Have installed jagasi and it registers with my asterisk server. But stuck from there!

kindly be informed that we need to do some customization for issabel call center module which is allowing logged agents to do outbound calls manually, please be informed that we want those manual outbound calls to appear on call center reports and recordings Skills: Linux, PHP, Software Architecture, VoIP, Asterisk PBX

I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk

We must have set up a new asterisk server that follows the normal asterisk guidelines. We also need a script where we provide the telephone number (DID), opening hours also agents who are queuing, then it must do the setup in configs.

I am looking for a developer to develop a telecom system and android and iOS apps.

We have an asterisk server at our HQ and we have sip phones in our branch offices. We want to Develop a SIP softphone that we customize with our brand and users can install in their windows, mac, ipad, android etc....We need to do voice and video calls with the softphone.....something like LinePhone and Zoiper.

1). Install 4 Asterisk instances Time for completion : 2 days 2). install and setup Kamailio on another isolated VPS instance; Kamailio will be setup as the registration server for all clients, and it will load balance all call requests on each registered Asterisk instance. Time for completion : 1 day 3). In conjunction with app developer, configure all the necessary dial-plans and trunks on the Asterisk instances and make sure they work with the database and scripts created by the developer to compute predefined data, and setup Asterisk to perform actions based on the computed results. Time for completion : When needed 4). Provide support in configuring SPA3102 devices (PSTN gateway/converter) Time for completion : When needed

I am looking into having my own miss call server/tools. I require assistance to: 1. Create / rent a server that meets the requirements needed to support sending miss calls to number from a local database. 2. Have a tool that sends ( dialer ) the outgoing calls 3. Setting up to be able to receive calls back from the numbers ...be able to receive calls back from the numbers called and redirect them to IVR or CC 4. Make a report at any point in time with current costs / income. I already have suppliers and trunks to connect to. Already have databases. I also accept consultancy to guide me in buying tools already on the market and putting everything together in a working manner. ( like what server to rent, installing asterisk PBX, setting up trunks, dialer, database which i can buy s...

I am looking for a developer to build a Asterisk pbx

Coding tips and live Double verification and more..

...rynek we współpracy z odpowiednimi działami w organizacji (R&D, Technical Support, Sales); • Współpracujesz z kluczowymi dostawcami oraz klientami firmy; Wymagane kompetencje: • Masz wyższe wykształcenie techniczne (preferowane elektronika, informatyka, telekomunikacja, elektrotechnika, automatyka); • Masz praktyczne doświadczenia z sieciami IP do aplikacji telekomunikacyjnych (m.in. VOiP, SIP, Asterisk, Trixbox, SNMP); • Komunikujesz się w języku angielskim; • Masz silną orientację na realizację celów; • Jesteś kreatywny, dobrze zorganizowany, samodzielny, jesteś nastawiony na współpracę; Nice to have: • Doświadczenie w pracy jako System Engineer / Product Owner; • Doświadczenie w pracy w metodologii Scru...

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

I have a system asterisk 3g modem, I need to install for me a database for IMEI of dongle purpose for this operation is when i put a dongle with new sim in my system dongle change automatically it imei. Pkease confirm it is possible to do it.