Sphinx asterisk jobs
I’m looking to setup a Hosted PBX company and would like help from a developer who has experience performing this service.
To consolidate our different projects, we decided to write our own backend service to Asterisk PBX by utilizing the AGI specification (see https://wiki.asterisk.org/wiki/display/AST/AGI+Commands ). Because of performance reasons, this back-end should be implemented by using plain C, with as less external libraries as possible. We aim to use this service on a broad range of hardware, so it is imperative that portability is provided. The only common denominator (for now) is Linux as a platform. Other platforms are not planned right now, but different architectures are. As a very minimal approach, x86, amd64 and arm (including aarch64) should be supported. As a first start, we would like to see a wrapper for every AGI command in .c/.h file so other functionality can be implemented on ...
kindly note we are call center company have UCCE and Asterisk systems and would like to integrate below systems with 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speech) with UCCE (Unified Cisco Communication Enterprise) and Asterisk 3- Integrate a ASR (automatic speech recognition) with UCCE (Unified Cisco Communication Enterprise) and Asterisk. 4- integrate Arabic AI solution with UCCE (Unified Cisco Communication Enterprise) and Asterisk.
kindly note we are call center company have UCCE and Asterisk systems and would like to integrate below systems with 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speech) with UCCE (Unified Cisco Communication Enterprise) and Asterisk 3- Integrate a ASR (automatic speech recognition) with UCCE (Unified Cisco Communication Enterprise) and Asterisk. 4- integrate Arabic AI solution with UCCE (Unified Cisco Communication Enterprise) and Asterisk.
Looking for a simple SIP dialer Mobile Application for iOS and Android which can register to our asterisk server and simply make and receive SIP calls. Having G729 codec enabled is preferred, otherwise GSM codec is required.
Need help with Vicibox. Have been getting some errors. There must be some mistake in the configuration I must have done. The calls go through fine, but when the customer disconnects, nothing is recognized in the asterisk.
Need help with Vicibox. Have been getting some errors. There must be some mistake in the configuration I must have done. The calls go through fine, but when the customer disconnects, nothing is recognized in the asterisk.
I am setting up a call center for my business. The call center agents will be making more outbound calls, hence the Autodial feature is critical. We plan to integrate Bitrix24 CRM and for Call Center Management as well. I need setup completed within 1 week, only experienced VOIP programmers with track record of executing similar jobs should contact me.
Hi I have small telco using bicomsystems PBXWare. I would like to get a whmcs module to integrate with pbxware. Would like the following: 1. PBXWare WHMCS Module • Create, Suspend /Un-suspend, & terminate account • Create SIP Extension (in Asterisk) • Use Customer phone number as CallerID • Send Welcome Email with SIP account detail 2. Module configurations: Configurations per WHMCS product • Configure initial balance upon account creation • Set max credit limit (Email will be sent upon limit reached) • Set Free, One time fee or recruiting fees • Change Caller ID • Set Invoice generation date 3. PBXWare > WMCS Invoicing • Auto create Post-paid payment invoices in WHMCS • Automatically adds 15 Day...
I need to create 4 example journeys. The server is FreePBX.
I’m looking for someone to setup an Autoscaling group of Asterisk Real-time Servers in AWS configured using CloudFormation and connected to an RDS Aurora database. Do you think you could help?
I’m looking for someone to setup an auto scaling group of asterisk real-time servers in AWS connected to a MySQL (Aurora) RDS instance.
I need some one to setup Multiple Telephone systems consisting of FreePBX , a Fxo Gateway Grandstream GS-GXW4108 , and multiple Voip Phones. You must have full knowledge and experience in FreePBX, asterisk and networking setup and must be able to continues support based on a monthly fee.
We’ve got an asterisk system that has some configuration issue with Twillio
i have 2 asterisk A B B is interconnected with voip provider i send calls from A to B and calls from A landing on B also go to same voip provider
Server B is interconnected with one voip provider ip2ip when we send calls from thats server to voip provider ip its go through Now i have server A i want send calls from server A to server B and from B that all calls which is coming server A ip forward to voipprovider ip Server A simple will create trunk and when all calls dialed from that trunk goes to server B ip need script which forward that all calls coming from server A ip forward to voipprovider IP
To fix bugs on an Asterisk-FreePBX Tel. System with WebRTC and Ubuntu OS. This project is long overdue and some backup help is urgently needed to get it going ! This is only for very serious professionals, who have the time to help out, on a complex Linux setup ! The project will be broken down into milestones for better handling. You must be able to work with Teamviewer !
Solve Asterisk Record problem - Record (,5,40,xk) recorda an empty file. We will give you access and you you help us resolve the issue.
Hi, i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
Hi Mohammed. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
Hi Ambiorix R. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
We need a person to help us with Vicidial/asterisk configuration. We want to upload a list, do automatic calls which will say: "Say yes to connect to an agent". If the person says "YES" the call is forwarded to agent's phone (through SIP/asterisk)
We are hiring a person to build and configure a asterisk project from zero. We are building a call center.
Hi i need one to install Asterisk freepbx on my server and do auto dailer to dail a special number by sip send call to ip i can set how call can done and how long of call and sperate time between each call maximum calls in same time 5 calls also set ivr on same server if any one from out side send call to ex 555 play ivr my own audio
We are looking for solution like a traditional GSM or CDMA VoiP gateway. This project will be separated in two par...Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This 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 to include option to show balance th...
We are using Bicom PBXware MT system and we are looking for someone to help our customers with tech support on live chat and phone, the system is very easy to learn and perfect English is required along with some experience with Asterisk/Hosted PBX
Hi usmanshery, I noticed your profile and would like to offer you my project. I would like to implement a C wrapper for Asterisk AGI functions. If all goes well, there may be more work to do as well. It is intended to implement several functions for an AGI interfacing with an Asterisk process.
To create a dialplan based on specified criteria for incoming calls to server. To add menu option on web interface. Offers between 100-150$.
Hi All, kinldy we need to hire a team to 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speach) with UCCE (Unified Cisco Communication Entrprise) and Asterisk 3- Integrate a ASR (automatic speech recogintion) with UCCE (Unified Cisco Communication Entrprise) and Asterisk Thank you.
...problem with server setup for Asterisk integration with Google API. We need to solve that problem. We started the project: By default, the project deploys all the necessary components, including asterisk 13. The only change we made is the channel driver, instead of pjsip we use sip. In the asterisk configuration files, we put the service key with the role of the DialogFlow API for connecting to DialogFlow services. The project contains this library Here you can compile a test client, in which if you insert the above key, they execute the DialogFlow requests. In asterisk, an error occurs during a call, presumably related to receiving authorization data: -- Executing [asterisk-irgkin@dialogflow-loop:2]
Hi, I would like to be able to write in freepbx outbound route (we have few) several numbers as optional caller-id's and the system will choose one randomly for each call using that route
... - Presence of unit, integration and acceptance tests - Use of class methods Results should include: - Documentation highlighting what the profiling tool does and how to execute it. It should also highlight its scope, scalability/limitations and future features that could be considered for development. - Documentation regarding func/class methods inputs/outputs should also be available in sphinx....
i have a asterisk openvpn server and my client connect server to puppy linux, i need develop my client system, puppy linux exchange TP-LINK router running OpenWRT, this vpn for voip business,
Need a linux expert to secure my goautodial server from DDos and asterisk attacks. I'm using goautodial 3.3 installed on centos 5.
I need a asterisk professional who already have the system ready.
We have installed open source VoIP platform Asterisk on our server. It's need to install to Asterisk speech driver for Google Dialogflow. It's need to connect simple Dialogflow bot to the Asterisk. Here is the driver code:
Setup my asterisk server for allowing WebRTC video calls
Hello. We need a person to help us with Vicidial/asterisk. Calls are not getting through, extensions are working and internal calls are working but inbound and outbound calls are not. SIP trunk is registered OK
Hello! We are a small fintech company based in Ukraine, bringing Fintech B2B solutions to global customers. As a top provider of B2B services, we help our customers make the decisions and follow the paths they need. With Blockchain, cry...especially situated to help our customers with their biggest business challenges. R&D team is concentrated in Kiev, and the company serves clients worldwide. We do qualitative software products to enable the business becoming more efficient and successful. We are looking for local VoIP providers or companies in Arabic region to start selling our products and provide customer support service. We use Asterisk (FreePBX) IP-PBX software calling huge amount of numbers worldwide. We will be more than happy to start cooperating with VoIP companies...
Hi i have Asterisk installed on raspberry pi all i want i want to allow calls from one public voip ip server with prefix go to extensions regards
Hello, We looking for freelancer who is expert in integration stuffs like twilio with this opensource java CRM - We need to also install twilio dialer and Asterisk server So we can achive the predictive dialer and auto dialer and SMS, whatsappp messaging - with two way communication, with the twilio integration as service provider. BId only if you have this all idea and can do fast. Thanks!
You need to write or provide a connector for integrating FreePBX 14 based on Asterisk 13 and Vtiger 7.1. It is also required to provide detailed instructions for installing and configuring the connector; after installation, Vtiger users must be able to make outgoing calls, receive calls, and listen to them.
Hi Yuriy M., I noticed your profile and would like to offer you my project. We can discuss any details over chat, i need you install for asterisk and free pbx or other solution for usb dongles for voip termination could u work on it and will need support as well after the project so what do you think ?
salam kamel, i'm from morocco as well, i need your support to install asterisk server with A2billing and gsm dongles, are u intrested ?
Hi cvierilviv, I noticed your profile and would like to offer you my project. We can discuss any details over chat, i need you install for asterisk and free pbx or other solution for usb dongles for voip termination could u work on it and will need support as well after the project so what do you think ?
We currently have a Sphinx implementation that is being used to process large amounts of data (140 GB in SQL databases and Sphinx indexes). One of our critical set of queries is no longer running optimally due to the large data set. We need some one with experience in SQL query optimization (indexing or procedure) and Sphinx query development (using the Sphinx API or query language) to help us optimize our queries for the application so that they are running quickly and not timing out (due to PHP timeout errors). This task will require: 1. Reviewing our existing queries 2. Optimizing them to leverage the Sphinx indexes more efficiently (essentially moving away from processing and pushing data via SQL entirely, or optimizing SQL queries s...
i need dongle expert who has knowledge in asterisk configuration, with dongle looking to solve an issue we are facing. The required should have already worked on configuring dongle with asterisk. Below is the log from asterisk CLI we are facing. This is first 3G modem we are trying to connect, we need to connect 3 more, in total 4 modems. WARNING[30258]: at_response.c:384 at_response_error: [card0] Error checking subscriber phone number -- [card0] Dongle needs to be reinitialized. The SIM card is not ready yet -- [card0] Error initializing Dongle WARNING[30259]: at_response.c:384 at_response_error: [card0] Error checking subscriber phone number
