Meetme asteriskpekerjaan

You`ll be creating a Bot on Python that shall automate the described procedure below for each Incoming Dialed number into a FreePBX/Asterisk softswitch. Call Flow: 1. Customer sends INVITE with Dialed number --> Asterisk AGI will then execute the Bot automating request (can be Flask App through a Curl call): 2. Python bot will send a POST request to a website with the Dialed number and 2 other values that I will describe in detail if interested in this project. (no login required) and fetch the result to either aprove or reject the call. 3. If number dialed got an specific result (out of 2 possible results) we tell the Bot to understand that specific result as a good number, therefore call must be allowed to continue through dialplan context. A few notes to have in consi...

I need an expert asterisk , vicidial about ivr outbound .

I need an expert for asterisk , vicidial , for my ivr outbound .

We need installation of Vicibox9 in proxmox, with 7 nodes in total 1 is web and database and rest 6 are dialers with asterisk 13 of vicibox, all boxes should be SHH seperately from outside all boxes should be port forwarded to webpage of vicibox all dialers communicate with public network with DB server IP address only

We need installation of Vicibox9 in proxmox, with 7 nodes in total 1 is web and database and rest 6 are dialers with asterisk 13 of vicibox, all boxes should be SHH seperately from outside all boxes should be port forwarded to webpage of vicibox all dialers communicate with public network with DB server IP address only

Hello we are looking for a team to implement WebRTC Plugin for Chrome to connect with our server and be able to use Call, SMS and video features, see call history etc

The Fourth Estate Public Benefit Corporation is a civil society organization with a mission to democratize the news for the public benefit. --- We are looking for a freelancer to setup, configure and secure an Asterisk server. We will be using XiVO, it's an Asterisk front end. The OS will likely be Debian. Documentation for XiVO is at: The system will need to be configured and connected to Twilio, our SIP provider and connected to two extensions that will be setup for soft-phones. The server will need to be secured. ---- The winning Freelancer could expect ongoing work to administer and maintain this installation

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

i need to integrate asterisk with php throw websocket

Buscamos un experto desarrollador en lenguage php y con conocimiento avanzado y experiencia en el crm VTIGER. El proyecto consistiría en : 1.- Proceso de instalación de un nuevo crm VTIGER 2.- Migración de otro crm, VTIGER a la nueva instalación. 3.-Actualización a la útima versión existente 4.-Integración a aplicaciones como mailchimp, asterisk y otras apps.

...flow and enhancements in our current implementation . We can roughly divide these functionalities into 3 parts: A. Webrtc Audio - Work required 1. Establish the webrtc currently implemented in view of scalability, robustness and voice quality and its integration with our application (we'll do the .net part but need help in the asterisk part of it). For this, as of no, we are just using one extension and there's no way to know who is making the call, so changes in asterisk context may be needed, to capture and keep the agent id and the no to dial from . Also, group audio conferencing and transferring the call to an app flow will be required 2. Configure webrtc for incoming calls too 3. Invoke the call pop up from the ivr and connect the caller to an agent 4. Enab...

I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer ph...someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not on Asterisk. 2. Instruction may instruct using CLI or exactly where and how to modify FreePBX files 3. Logic of FreePBX related with these 2 tasks 4. Project will be considered as completed only after instruction (PoC) will be tested on FreePBX and con...

I want to be able to stream music for agents, from our PBX, to replace MoH files, played by Asterisk. The music will be for entertainment and make sure that agents are connected to the PBX. There may be different ways to accomplish this. Please also see attachment.

I installed Vicibox9 on 5 different servers, 3 Dell Poweredge 2950 Servers, 2 i5 Desktops. 1 Dell keeps getting time sync errors even though the time is fine, 2nd dell the asterisk keeps crashing every couple of mins, third dell the time just went blank and wont sync at all, the i5 desktops (Clustered) works for awhile then also gets time sync errors, then I need to reboot them after which I sync the time manually with yast then they come back online again. We need one solid working vicidial server working. Dell Poweredge 2950 Servers All: 2 X Intel(R) Xeon(R) CPU E5405 @ 2.00GHz Quad Core 8GB Ram Raid HDDs Dell1,2 HDD 1TB Raid 5 Dell3 HDD 160GB Raid 1+0 Internet Connection 2 X 10Mb Fibre lines

I need an Android app. I would like it designed and built. Something similar to the meetme app...

Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.

Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will not be considered.

...have no parameters and one will have the name, password, and the amount of money parameters. (2 points) 12. The password secret encryption algorithm is sequenced below: (10 points) 1. The first two characters will be moved to the end of the string of characters. 2. A random number greater than 0 and less than 10 will be inserted between the 2 and third characters in the string 3. An asterisk ( * ) will be place after the 7th character 4. The first character of the Players’ name will be the last character in the password. 13. The password is stored and displayed as encrypted. The Player class will also have a method to display the decrypted password. (10 points) 14. Add and use the Console class from chapter 7 or a modified version to validate all user input ...

Hello, we need asterisk AMI script (syntax) for yeastar PBX we need functions must work via triggering AMI commands described and tested : 1. hangup 2. mute 3. attended transfer 4. hold please only serious freelancers with experience. Please be aware that Yeastar PBX has limited manager commands

...Trying to update: Unsupported Version of Asterisk, You need at least 11.11.0 you have 11.8.1 Running Amportal: amportal a ma refreshsignatures Fetching FreePBX settings with gen_amp_conf.php.. /usr/local/sbin/amportal: line 52: export: `®': not a valid identifier /var/lib/asterisk/bin/freepbx_engine: line 100: export: `®': not a valid identifier Getting Data from Online Server...Done Checking Signatures of Modules... Checking announcement...Signature Invalid Refreshing announcement Downloading 42672 of 42672 (100%) The following error(s) occured: - File Integrity failed for /var/www/html/admin/modules/_cache/ - aborting (GPG Verify File check failed) Trying to update the key: [root@hfp ~]# sudo -u asterisk gpg --refresh-ke...

I have an Asterisk-FreePBX System with multiple disks that needs some fixes. If You are a specialist in this field, lets talk. --- This is not for people who think all the answers are on the Internet ! This is for experienced specialists. Requirements: Asterisk, FreePBX, SSL Certificates (Letsencrypte), Apache, multiple disks on system, web dev, PHP, etc. Will divide into milestones.

...following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real Time Control Protocol. (3) Programming Languages: (3)(1) C/C++. (3)(2) Java. (3)(3) Swift/Cocoa/Cocoa Touch. (3)(4) Linux Debian Application Programming Interface. (3)(5) Android APP Development. (3)(6) iOS APP Development. (3)(7) HTML & JavaScript & MySQL & PHP. (4) OpenSource Softwares: (4)(1) GnuGK. (4)(2) Free SIP APP's( Android, iOS ) Source Codes. (4)(3) OpenSER. (4)(4) Asterisk. (4)(5) Speech To Text. (4)(6) WordPress or Any Other CMS Software. If yo...

Hello, I have an asterisk PBX vers 11.22.0 . I am using a Polycom sound point IP 650. All works fine except for the transfer button. The transfer on polycom use SIP REFER to transfer the call. This is not working. Need help from anyone who know about the subject. Please respond to this project with "What up Dingo" at the beginning of your message so that I know you have read.

Hello I have Asterisk dialer and I need to set up speech to text transcription (ONLY) I use to use to use IBM watson api for this, but it has become too pricey. it is 1 Min length audio of ivr recordings each. But total millions of files. every 2-3 months 7 million 1MB, 1 Min audio files.

Hi We are looking for a freelancer experienced in Asterisk. Current developer works at another job, so you will work with me for a long term if you want. hourly rate is 25~35. 40 hours per week Thanks Anthony

This is an on-going project with various tasks managing asterisk Please apply only if you have experience in this.

I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer ph...someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not on Asterisk. 2. Instruction may instruct using CLI or exactly where and how to modify FreePBX files 3. Logic of FreePBX related with these 2 tasks 4. Project will be considered as completed only after instruction (PoC) will be tested on FreePBX and con...

Actualmente tenemos ya funcionando una centralita Asterisk con FreePBX y queremos implementar un sistema de encuestas telefónicas para valorar la atención telefónica que damos a nuestros clientes. El funcionamiento deseado podría ser UNA de estas 2 opciones: - Opción 1: tras hablar con uno de nuestros agentes, la llamada pasa automáticamente a una IVR para realizar la encuesta. No nos sería válido que se deba hacer el desvío a la IVR manualmente. Se debería avisar al inicio de la llamada que se va a hacer una encuesta. - Opción 2: Tras finalizar la llamada, al cabo de pocos segundos/minutos se hiciese una llamada saliente con la encuesta. En cualquiera de los 2 casos, los requisitos serían...

We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a public IP (direct). Endpoints are Acrobits sofpthone users (android/iOS) connecting via WAN. there is nothing in between. All end-users use TLS+SRTP. and Acrobits Push. So ping is huge sometimes. I suppose qualify option may be the cause here. Official FreePBX forum treads ignore the issue and ask to order their paid support. As the issue is in the production system, there is no place for exp...

Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database

Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall VPN Client

Looking for Asterisk, FreePBX with WebRTC specialist to verify and fix existing PBX. This is no place for amateurs !

I have a running asterisk PBX - i will to rebuild new one with asterisk. i use Asterisk API, Databases.

Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action

migration of asterisk to clean server

migration and debugging asterisk

migration to asterisk 16 some scripts to get up and running

I am migrating my asterisk server to latest . I need some help to resolve the issues

we need an expert in Networking to create a VPN between 2 servers 1. both servers are a VM in a windows 2. Server #1 we have asterisk that need to allocate an Public ip #2 is connect to local GSM Gateway to test it we need to make a call via the asterisk using the vpn and ending up at the gsm gateway

Perfil profesional: Informático, Programador, Ingeniero Telecomunicaciones, Matemático Conocimientos técnicos (Experiencia mínima 5 años en cada una): - JAVA - PHP - JAVASCRIPT - LINUX - Gestion de sistemas y redes - Bases de Datos (MySQL y SQL) - Experiencia en servidores Otros conocimientos técnicos: - ERPS y CRM - Android o IOS - Asterisk y/o FreePbx - WEbRTC -Idioma Español nativo Habilidades Blandas - Resolutivo - Dinámico - Responsable - Pro activa - Habilidades de liderazgo - Toma de decisiones asertivas - Excelentes relaciones interpersonales ***Se requiere disponibilidad inmediata a tiempo completo. SI NO CUMPLE CON LOS REQUISITOS FAVOR NO APLIQUE

Resolve issue with “Cannot Conner to Asterisk” error. Update server and enhance security.

I need someone to help configure the Patton SmartNode 4114 (FXO Port) with Freepbx/Asterisk so we can use SmartNode 4114 as a VoIP gateway for PSTN lines.

we need to get support for any one who know the chan_dongle and asterisk good

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