Asterisk voicexmltrabajos

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: I may probably also need help with G729 codec configuration

I want to build an optimized image of the latest version of Asterisk for the latest Alpine Linux stable. The final product is the Asterisk compiled and tested with load in an AWS server and Docker. The system should have the codecs g729, opus and SILK compiled. The AWS image should be able to run without docker The Sources should be available to recompile minor updates

i need some one to help me set up dial pattern for my voip , i have to outbound routes one for international, and one for local just need someone who understand how to set it up correclty I am using vitalpbx (asterisk)

I need Odoo developer familiar with Asterisk distribution like Issabel , freepbx ...etc .

...Internet by our clients, changed to pdf again or to png and send by email, as invoices or Credit Notes, to their final destination. The client must never see the spreadsheet, he only sees the background image with the data he entered into the Cells ! Min. knowledge: 1) LibreOffice Calc Spreadsheets. 2) Software (open source) to change the spreadsheet files to pdf, without loosing calculations ! 3) Asterisk PBX would help This is a none standard project and absolutely not for beginners !! Very good knowledge of the way Spreadsheets and Billing software work, is needed. We have a Multi Disk Server with KVM disk management and MySQL DB ready to upload. -- Good and quick communications are essential ! You must have a computer with a Linux OS --- This is not for Windows !! Pleas...

Hi all, I am looking for a system admin to install A2Billing on my CentOS 7...work: I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,exten,priority=a2billing-did,s,1 I have tried this installation guide and some others on 2 different servers: one with Asterisk 16 and FreePBX 15, and the other with Asterisk 13 and FreePBX 14. I'm looking for an engineer with A2Billing experience so that to work on this task of installing A2Billing. It will be for the engineer to choose the server to use from the two I have mentioned above. Kindly get back to me as soon as possible, as this task is quick urgent.

We would like to connect one Microsoft Teams user to a DID on an Asterisk box via SIP

Freepbx 15 installation on DEBIAN or CENTOS OS. I tried many times but it never worked (calls drop after 32 seconds, or I hear nothing) I even would like to use API REST or other module to call url when incoming call to specific number, then this url respond with number that asterisk needs to callback.

hello, we are looking to implement AMD on asterisk using give below case 1- We will send call from freeswitch to asterisk 2- Asterisk will send call to route. 3- once call is answered by called party 4- asterisk will run AMD 5- if AMD is detected the asterisk will hangup 6- if live person is detected then continue the call Thanks

Ciao Leandro, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

Ciao Daniele, ho visto il tuo profilo e vorrei sapere se sei interessato ad un progetto di integrazione sw con Asterisk per fare telefonate con riscrittura del caller-id. Saluti, Leonardo

Hello, I want to build API [Java or php or C/C++] to call to Asterisk Server by using Android also from Windows [Browser]. The API will be one for both Android as well as Windows browser. I have purchased Asterisk SIP server users [Two number] and they gave me two extensions alongwith Server IP and password. Will shore the details after award the project What I need a API to integrate with Android as well as call from browser. The requirements: 1. Need API to Generate a call and receive a call [Both from Android and Windows Browser]. 2. Call recording 3. call Duration Log 4. Source Code after checking the Demo.

I have a set of c# asterisk applications compiled with Visual Studio that communicate using TLS 1.0 that must be recompiled to be deployed using TLS 1.2 and TLS 1.3. I am looking for and experienced Asterisk and C# programer to compile and assist in testing the deployment.

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...

I have a working Asterisk 13 phone system, but my Trunk provider (Flowroute) recently retired their POPs and forced everyone to use PJSIP vs SIP protocol. This has literally shut down my phone system. I tried to build a new system from scratch with the latest Asterisk 13 source code, moved over my dialplans to it, etc. but now I can't get any luck with authentication between both my phones and my Asterisk server and the trunk provider. I see traffic coming from there successfully when a call is made, but it is being rejected with a "404 Not Authenticated" response by my Asterisk server. If you are a seasoned Asterisk expert, specifically with experience using Flowroute, this is probably a simple support job. If you are not, please do n...

Павел привет. Куда я могу тебе написать, ищу человека на проект по телефонии Asterisk

Hi, We are looking for a consultant to assist in setting up a Kamailio SBC to integrate Teams & Asterisk. Brief overview of requirements: - Centos 8 - Kamailio - Support for multiple MS Teams instances - Support for multiple Asterisk servers - Certain MS Teams instances need to be linked to specific Asterisk servers Required: previous experience integrating Kamailio/opensips with Teams This job is urgent. We need quotes today to be signed off tomorrow (Thusday) and installed Friday. Thanks, Carel

Hi, Need an Asterisk Expert to do one or 2 little task. If you can work fast and reliable price then can offer more works. I will pay per task. So please provide your price as per task. First I need to record the VOIP Calls. Please provide your price for this one. Please mention "calls" at the top of your bid so I will understand you've read my description clearly. Otherwise I will ignore your bid. My budget is $50 for this project. Thank you.

Hi there, I have a freepbx solution connected to PRI line. When I have a low number of concurrent calls, everything is fine. But when the number of concurrent calls increases, calls start to drop. I need a super experienced asterisk to help out.

Build kamailio server with the following configuration: Transparent proxy to Asterisk server Multiple registrations permitted per account User agent whitelist

Владимир привет. Я ищу человека на проектную работу по разработке ПО для обработки звонков Asterisk. Я по адресу?

Привет, как я понял Павел. Я ищу человека на проектную работу по разработке ПО для обработки звонков Asterisk. Я по адресу?

Using asterisk 12.6.0 / FreePBX 12.0.1rc29 I will not be able to provide server access. You must use your own server for testing. Part 1: 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,...

Hi, We are looking to integrate Odoo v12 or v13 with Asterisk on our server in order to setup a small call center mostly for inbound calls. There are some links on Odoo documentation but we need someone that can show us a demo that works. We prefer to work per fix price at this moment so feel free to bid your price in order to make this process up and running. Note: Please answer to this question for a successful bid: Can you show a functional demo of a similar project you did in the past? If yes send us details on how your demo can be viewed.

professional with great experience in Asterisk, PABX, predictive dialer, softphone and wertc

I am not strong on Python and need help (share screen and give control to VM) Environment: Ubuntu 20 LTS, Python 3.8, Asterisk v16 Note: 1. Asterisk (works OK - Inbound call, plays monkeys via a context with an external softphone) GOAL: 1. A "hello world" python to play the "monkeys" prompt on an Inbound call using Asterisk ARI REST API. 1. Using this Python module ARI 2. See "Hello world example" My basic issue/problem >>> import ari Traceback (most recent call last): File "<stdin>", line 1, in <module> File "<frozen zipimport>", line 259, in load_module File "/usr/local/lib/python3.8/dist-packages/", line 8, in <module> File

...and for that we are using a payment interface named cashfree. So integrating this with the already finished store and completing whatever is left is the work. This also include the basic interface designs and tweaks. You can understand the working of the offer zone from the block diagram which is attached. The key features are listed below: (Features not completed in the website are marked with asterisk) User (Website and Android) ● User login/Sign Up with OTP ● Search tool ● Integration with Cashfree and Payout feature* ● Product List/ filter ● Product Collections ● View product info/categories ● Apply coupon code ● Checkout Cart ● Tax adding* ● Check out multiple products, Shipping Address & Billing Address. ● Product details page- Images, Description, Related Products ● S...

i have a goip configured with asterisk13 on ubuntu, incoming calls stopped working , need help to fix it , i can pay 10$ fro this help, please do not bid high amounts i will not entertain high bids.

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures:

We need someone to take care of an Asterisk installation. It's running on a rather outdated server, should be upgraded to something newer eventually.

need to Asterisk and Issabel expert to install and customize on my local system.

need to Asterisk and Issabel expert to install and customize on my local system.

to do: dialplan, sip-trunks, security.

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...

...spreadsheet, he only sees the background image with the data he entered into the Cells ! Min. Requirements: 1) LibreOffice Calc Spreadsheets. 2) Software (open source) to change the spreadsheet files to pdf, without loosing calculations ! 3) Open source Software to connect Asterisk to Cells on the spreadsheets. N.B.: Server and Asterisk PBX are on different networks ! This is a none existing, none standard project and not for beginners !! Very good knowledge of the way Spreadsheets, Billing software, Credit Notes and Asterisk PBX work, is needed. We have a Multi Disk Server with KVM disk management and MySQL DB ready to upload. -- Good and quick communications are essential ! You must have a computer with a Linux OS --- This is not for Windows !! Please read ...

basic config, incoming call, outgoing call, extension, ivr.

i need one who have good experiance on asterisk to solve some issue for me

Responsibilities Handling Linux Server Install Switches, PBX on Servers Maintain VoIP Billing Providing Support Server Error Handling Write Custom Codes for Development Requirements Minimum 1 Year experience in VoIP Good Knowledge of Freeswitch and Asterisk - working on astpp billing, fusionpbx etc. Good knowledge of mobile dialers an advantage too Well knowledge of SIP Servers Good in development of AGI,FastAGI, Dialplan, FS Scripts Knowledge of JavaScripts and Lua. Working Knowledge of LAMP Full time online position Educational Qualification - MCA, B-Tech,

...sees the spreadsheet, he only sees the background image with his entries into the fillable fields ! Requirements: 1) LibreOffice Calc Spreadsheets. 2) Software (open source) to change the spreadsheet files to pdf, without loosing calculations ! 3) Software to connect Asterisk ( CEL - not CRM !) to fillable fields on spreadsheet. N.B.: Our Multi-disk Server and Asterisk PBX are on different networks ! This is not a standard project and not for beginners ! - Very good knowledge of the way Billing software, spreadsheets and Asterisk PBX work is needed. We have a Multi Disk Server with KVM disk management and MySQL DB. -- Good and quick communications are essential ! This is a Linux only project --- No Windows ! ***** Please also read and understand the attachment ...

Asterisk AMD modification. Want to increase answering machine detection time. If machine is detected the system should prolong the call.

I need custom the billing report of Issabel 4.0 Asterisk 11. Existing report is already fixed the column, so i need to add additional column (around 2 or 3) and also parameter menu button that to be add to this column. The parameter something like Cost Center, Employee ID, etc. Thank you.

hello I will offer remote desktop to configure a freepbx asterisk install. you have to add a second ntw card , do static routing and integrate sip trunk with my telephony provider

Hello, we have found a bug in asterisk 11.22.0 where we have also found a fix. The fix can be found here We need somone that can make the changes to asterisk source files, recompile asterisk to make it work , and document each step so we can duplicate. Please respond to this project with " I can recompile asterisk and fix this bug" at the beginning of your big so that I know you have read.

setup the SIP on freePBX and asterisk IVR managing call routes etc

I need a script in angular which in can make click-to-call option. The Call flow as under: 1. User click the Click-2-Call button 2. Asterisk make outbound calls to two mobile number. 3. When phone is answered by both parties, asterisk bridge the calls,. Require DEMO

i hve a server with asterisk incoming call to an ivr the ivr has stages every stage has a task play audio to the caller, msg / menu / data from a record in a db / get data from callers telephone keys / gething a audio recording from the caller, i need som chainges in the scripts and som new scripts. i need som new tables in the db and to edit som tables. i need someone how can lern and work very fast. i need to talk to you in english , and show you online and work whit you so i can test every step you do and get it redy fast. only if you think you do it my way, bid for

I have already done AWS Asterisk Instance with FreePBX. I needs Session Border Control with Direct Routing to Microsoft Teams. I need this done in one day. I need to see the work done using Team Viewer. I need an understanding of how you plan to take on the challenge and what SBC you would use. Looking for only experienced AWS Instance, Asterisk, FreePBX, Session Border Control, and Microsoft Teams.

I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice vers General Deliveries The application working in APK format Full source code Simple manu...

Good night, I was searching for poll solutions, and I saw that you offered to perform a service, I am interested in a similar tool. do you have a tool or disoniblity to create it? my center is Issabel / Elastix project link where I saw your contact. https://www.br.freelancer.com/projects/php/pesquisa-satisfa-elastix-asterisk/?ngsw-bypass=&w=f

Hi Abid Ur R., I have an Asterisk server I have had running for yeas. I switched trunk providers to and now my text(they have mms now if you didn't know?), in nor out isn't setup correctly and doesn't send to any of my linphone, csip or hardware grandstream phone setups. Do you know how to get my asterisk server written properly to send and receive text(if mms now as well, I'd pay more)with voip.ms. If so, what's your lowest rate and how long needed with ssh access? Paul