Nagios asterisk project jobs
I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls using his browser. The gateway will have SQL db that contain the user credentials to connect and the SIP credentials to register to the asterisk server
We’ve got an asterisk system with two trunks. We need some extension configuration changes and some ongoing support
I want to change A2billing AGI to FastAGI due to performance and scalability. I need a very experienced person with a2billing and ofcourse asterisk.
I wrote a Script that returns a True or False boolean according to a lookup number from a website. I need someone to write an AGI on my on my Asterisk server, whose main purpose is to forward a Dialed Number to my Script in Python, and afterwards if the number dialed is returned with True, then the call should be allowed to go through. If the number is returned with False, a 503 svc unavailable must be returned to Originator. You can reach me to discuss further aspects of this project.
hello, we use a "less secure app" with our Asterisk PBX voicemail to email message notifications using gmail. gmail wants users who use this option to make them more secure. if you google "less secure apps" and gmail / gsuite you will see what needs to be done. this is what needs to be done: The G Suite Team <gsuite-noreply@> Tue, Jul 30, 12:16 AM to me G Suite logo Hello Administrator, We’re writing to let you know that on October 30, 2019, we’ll begin removing the setting to “Enforce access to less secure apps for all users” from the Google Admin console. This setting will disappear from your Admin console by the end of year. Removing this setting will help keep your users’ accounts secure, as access to less secure apps (L...
Convert files from wav to mp3 files after a call is made, historic data and new data automatically after a call is made. when change is made I want to see and hear from crd reports. review and clean log data from /var/log/asterisk make rule to minimize log file size /var/log/asterisk/fail2ban /var/lib/fail2ban /var/lib/fail2ban/
We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document
we need an expert in call termination that have expertise in 1. DINSTAR equipment VPN 3. and avoiding DPI 4 asterisk we need to set up a rout
I started to build this web based SIP phone using - - but other work has left the project incomplete. I need the project completed and updated to use the latest version of , 0.15.6. You will be provided with FTP access to the current source files including the HTML, CSS and current JS files. Additionally 3 SIP accounts will be made available for testing as you progress, these are own our own SIP server running Asterisk 16 with the PJSIP stack. The current version allows for successful calls to made, both inbound and outbound, placing calls on hold and resuming those calls and call muting. The biggest things that needs to be done are call transferring, both attended and blind transferring and conference calling. I have BLF (Presence) working OK and the web
We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document
We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document
Hi Ibrahim Ali M A., I noticed your an expert in VOIP and asterisk. We are having issues with our VOIP system - in particular outgoing calls through SIP trunk are getting cut off in 6 minute 39 seconds. Asrerisk server running on CentOS. Can give access through SSH.
HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK someone who can teach.
HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK someone who can teach.
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 parts. One is mobile application and another one is registration server. The 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 ass...
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
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.
