Asterisk elastixtrabajos

I registered for an online sip provider. When I use their app, I can add caller-id's by verifying phone numbers by sms. Without verification, changing to any custom phone number, is not possible. When I use the same SIP from my own asterisk server. I am able to change the parameter CALLERID(num) to ANY value, and without verification, the caller ID shows up on calls. I would like to use the provider directly, without integrating it into my own asterisk server. I have an open source iOS app = linphone. I can set custom Headers etc.., but I have not found a way to set a custom unverified number as the caller ID. Objective: Provide information which will allow me to call from any caller ID using the open source linphone swift iOS app.

I need integration of opensip over asterisk. NOTE :- DONOT BID UNTIL YOU DONOT KNOW ABOUT OPENSIPS

We are looking for someone who have a sound experience of Asterisk and Json. He or She must possessed some sound quality which should adhere to our standard

hello I have asterisk ip pbx and i need someone to setup call us web rtc button on our website please see the image so you can understand our requirements

We need someone with a lot of knowledge of asterisk and webrtc. We are looking for a partner who can handle the ongoing development and error correction. We are only looking for people where we can see great reference on the asterisk

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which will be played randomly) if not allocated to any client. -Numbers page show all det...pay clients for their incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" includin...

Integrate my FreePBX system with Zoho CRM - Phone Bridge See more: zoho c, zoho, bridge, asterisk pbx system, pbx phone system, crm pbx, teamspeak phone bridge, crm zoho user guide, crm zoho magento, asterisk phone system, integrate crm asterisk, adobe connect phone bridge, php crm zoho, freepbx phone, asterisk pbx crm, sugar crm zoho crm, integrate asterisk pbx, phone freepbx, asterisk bridge, a2billing freepbx integrate

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

I want to learn opensip integration in asterisk . I already install asterisk server . Thanks in Advance

I want to install opensip in asterisk... Note :- Donot waste your bid until you donot know about opensips

This project is to write instructions to make special Templates with Scribus, to be used by total newbies. The Templates use a background image as base, with sometimes a spreadsheet for Cell addresses, used by Asterisk PBX.. Knowledge required: Scribus, LibreOffice, Asterisk, Java Script, etc. See attached Start Page. Details to be discussed. NOT FOR BEGINNERS !

I have elastix 4. I need call popup. I will configure myself. Just share the code. I will apply in my PBX. if it works, you can get your bid price.

I need to configure Kamailo SBC on Ubuntu 18.04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usab...04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usable and repeatable documentation for: - Install - Configuration of Kamailio - TCP / UDP / WebRTC Proxy passing through to Asterisk on Private IP - Registration pass through to Asterisk with rewrite to present sourceip as reg contact on ast. - 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 MS Teams 3 <--> Kamailio SBC <--> Asterisk ...

Hi Alexander, I noticed your profile (VOIP Asterisk freepbx) and would like to offer you my project. We have a conference call going via an Asterisk server with (let's say) 100 users. We need to record individual users in 15 sec intervals -> send wav files to Google for speech recognition -> process the returned text by searching for a few bad words only -> if at least one bad word is found then write to an internal db and trigger a mute function on that user. Is that something you would be interested in? If yes - how much would you charge and how long would this take? Thank you, Michael

Need an Outlook 2016 plugin which allows for CTI in an asterisk based call center

I just need someone to immediately (1-2 hours) install Elastix pbx and whatever it comes with or is necessary to work on my VPS.

I have Asterisk dialer for outbound shortcalls compagins its dialing perfect all calls issue what we need to fix which see we are getting if we are sending 10calls per second it will dial for a min then pause for 50sec then restart dialing i want keep sending calls at 10cps without take pause please bid only if you understand issue we both can save time of each other

I need a Dashboard for Asterisk PBX to show: - Number of calls in que (ringing) - Number of extensions busy - Total calls answered - Total not answered must be nice GUI and live auto updated

Using asterisk 12.6.0 / FreePBX 12.0.1rc29 I need a method to add a leg to a call through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script. For example, 1231231212 from the outside calls inward, it rings through a ring group, and ext 500 picks up. I want to have a button on my crm call a php script to transfer the call that is ongoing between the outside number and ext 500, to another internal ext 100, or even external phone number. By the end of the function being completed, there will be a joined called from ext 500, the external number, and ext 100. The user at ext 500 will end the call by hanging up, while the call between the external number and the ext 100 continues. In some types of failure, a ...

Looking for a developer familiar with Asterisk Opensource PBX software, to assist in a build out of a multi-tenant PBX on that platform. Please apply with your experience with Asterisk, and availability.

I need to schedule a cron job to export a CDR report. I am using FreePBX, Asterisk, Ubuntu and AWS. Here is a reference link: I need this done now using Team Viewer so I can learn and record the screen.

My customer need integration between IVR Asterisk (Issabel 4.0) and external Database. When the call arrived at Asteriks then IVR will picke-up that call, then play greeting to enter the ID or something. Then Asterisk will sen the ID to external database. Database will search the ID in it's database, when match then will send back to IVR to prompt information text to speech to caller. Thank you.

need to adjust calls per sec secript for asterisk for outbound calls can you able to do right now i have runing traffic and want do now fix that iissue my asterisk sending too much calls want to know if need to send 12-15 calls per sec for dial outbound how can fix

Using Asterisk 16 API - Open mysql database: Place call : If no answer update database row with status - done: If answer play mp3: wait for touch tone response- update database row with status - play text to speech from row- done: Languages can be c,c++,php,python

Caut o persoana care sa ajute la instalarea si configurarea unui server FreePBX Asterisk, un server VOIP. Avem deja un calculator blocat pentru acest lucru.

Hi Amal, I need to install asterisk server and setup intercoms between 100 extensions. There will be sip trunk which I need to configure so that outgoing call will go from that trunk. There are few more details I would like to understand like feasibility of the solution. Let's connect to understand it further.

the required tasks : 1- Implement Media Server using UniMRCP 2- Integrate Google TTS (text to speech) with UCCE (Unified Cisco Communication Enterprise) and Asterisk 3- Integrate Google SR (automatic speech recognition) with UCCE (Unified Cisco Communication Enterprise) and Asterisk

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.

jasa remote setting untuk keperluan kami yaitu: Menghubungkan antara GOIP dengan elastix dan goautodial saat ini saya memakai linux elastix dan aplikasi goautodial serta GOIP(gateway voip)

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. ...