Asterisk adhersionpekerjaan
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 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
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
... - - 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 phone currently sends AJAX calls to a CGI script for
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
Instalar asterisk y configurarlo para conectar con odoo
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.
Hace un mes me crearon un panel que puedo acceder por web y junto con Issabel Asterisk me registran las llamadas que se hace y lo muestra por la web. Hace unas semanas todo funcionaba perfectamente al 100%, al cabo de 2 días me dio el mismo fallo que muestro aquí, y el creador me lo arreglo, pero al cabo de unos 2-4 días dejo de funcionar de nuevo, y ahora me dan largas y no lo quieren arreglar. >>>El procedimiento es: En la web doy de alta teléfono A. Desde teléfono A realizo una llamada al teléfono de Test (éste siempre es el mismo). En la web me muestra el resultado de la llamada del teléfono A registrando fecha y hora. Y así sucesivamente con todos los teléfonos que de alta. >>>Lo que me...
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.
Create 20 video call accounts in my Asterisk server
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.