Sipx asterisk elastixtrabajos

We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server

VoIP developer requires good knowledge of voip development free switch and asterisk developer .

Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash drive, and those USB flash drive will be delivered to our Server B type client (ther termination provider) A. Any mini Linux distribution exam- puppy Linux , linux mint B. Fedora desktop distribution C. Centos 5.8 or 6 7. Server A to Server B voice traffic wi...

Asterisk 17.4 is downloaded and extracted, but needs to be properly installed onto a server for us. PJSip config to a SIP provider, the ability to provider music/advertising on-hold, a couple simple queues. We do not want freepbx, please do not suggest it. Just install asterisk and, as needed, help get config files setup.

Both Chrome extension and windows application will provide connection to Asterisk AMI for click2call dial and popup based the caller callID. The chrome extension will change all page phone numbers (based on a phone pattern set in option) to click to call link with a phone display. The extension should provide same capabilities as FOP2 The windows application will provide DLL for other application to get and set events.

1. There are multiple asterisk servers, each provide services to multiple tenants. 2. From a main centralized server (panel server), I need the option to load file (word, pdf, tiff, txt) and send it to the relevant asterisk server. 3. Current asterisk servers has a local webpage to load and send fax. 4. panel server has full access to local asterisk servers and open AMI connection.

i did install the freebpx but i have nothing but problems, i need someone to help me with it, secure it and make sure its working to its optimal condition. .

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

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.

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

Hello, I looking for someone to help us to build Asterisk IVR/PBX with call blast dialer and predictive dialer.

Do NOT ask me budget. just bid what ever that you want to get paid. Attached requirements. first app we have today on 3cx app. we can give to show (3cx flow builder) now we need to crete on asterisk. need to read and see if understand all well.

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 ...