The document discusses using Asterisk Gateway Interface (AGI) and Asterisk REST Interface (ARI) to connect multiple Asterisk instances in a large scale environment. It begins with an overview of AGI versus ARI, explaining that AGI interacts with the Asterisk dialplan directly through applications while ARI provides a RESTful API. The document then covers using both together by having an AGI script select a ARI application. It also discusses handling calls across multiple Asterisk instances using an ARI proxy like CyCoreSystems' or retel-io's to provide a single point of control.
This document discusses using Asterisk Gateway Interface (AGI), Asterisk Manager Interface (AMI), and Asterisk REST Interface (ARI) to build telephony features. It proposes using AGI for generic call setup/teardown and feature determination, and ARI apps running on an ARI proxy to control individual features during calls. This decouples features from Asterisk and isolates them, improving testability, replaceability, and avoiding direct developer interaction with Asterisk dialplan. The approach moves routing logic away from Asterisk and enables any feature via a low-level ARI API and high-level feature control.
Este documento describe las nuevas interfaces de programación de aplicaciones (APIs) de Asterisk, incluyendo ARI (Asterisk Restful Interface). ARI proporciona una interfaz RESTful, transmisión de eventos a través de WebSockets y la aplicación Stasis para comunicarse con módulos de Asterisk. Anteriormente, las primeras APIs de Asterisk fueron AGI para scripts y AMI para enviar comandos y recibir eventos a través de sockets. ARI representa la evolución hacia una interfaz más moderna y basada en REST para Asterisk.
This document summarizes Pascal Cadotte Michaud's presentation on getting a live transcript of calls from Asterisk. It discusses three main ways to stream audio out of Asterisk: RES_ARI_STREAM, Asterisk's external media functionality, and AudioSocket. RES_ARI_STREAM uses the Asterisk ARI WebSocket to stream audio. External media allows Asterisk to send RTP to an external host and port. AudioSocket is a simple TCP-based protocol for streaming audio. The document provides code examples and links to demonstrate each method. It encourages connecting with the Wazo Platform team to learn more about building solutions using open source telecom components.
This document discusses how to configure Kamailio as a session border controller (SBC) using Docker. It provides an introduction to Kamailio and explains how to install Kamailio from source on a CentOS Docker container. Specific steps are outlined to configure Kamailio to use MySQL, create the database schema, modify the configuration file to use the private and public IP addresses, and start the Kamailio server. The document also provides instructions for testing Kamailio by getting the error log, creating SIP user accounts, and registering a SIP softphone.
This document summarizes a presentation about Kamailio. Kamailio is an open source SIP server that can function as a proxy, registrar, location server, application server, dispatcher and websocket server. It has a modular architecture that allows additional functionality through modules. The presentation covered what's new in Kamailio 5.4 like new modules for Kafka, MQTT, dialog tracking and STIR/SHAKEN, as well as common deployment scenarios and examples of Kamailio configurations.
Kamailio combined with Asterisk creates and incredibly robust and durable VoIP framework. With scalability and security, adding Kamailio to an asterisk deployment makes sense and saves money.
This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.
Using Asterisk and Kamailio for Reliable, Scalable and Secure Communication S...Fred Posner
Presentation from AsteriskWorld 2017 at ITEXPO. Discussion of how I started with Asterisk and Kamailio as well as how to build Reliability, Scalability, and Security into your telephony platform.
This document discusses using Asterisk Gateway Interface (AGI), Asterisk Manager Interface (AMI), and Asterisk REST Interface (ARI) to build telephony features. It proposes using AGI for generic call setup/teardown and feature determination, and ARI apps running on an ARI proxy to control individual features during calls. This decouples features from Asterisk and isolates them, improving testability, replaceability, and avoiding direct developer interaction with Asterisk dialplan. The approach moves routing logic away from Asterisk and enables any feature via a low-level ARI API and high-level feature control.
Este documento describe las nuevas interfaces de programación de aplicaciones (APIs) de Asterisk, incluyendo ARI (Asterisk Restful Interface). ARI proporciona una interfaz RESTful, transmisión de eventos a través de WebSockets y la aplicación Stasis para comunicarse con módulos de Asterisk. Anteriormente, las primeras APIs de Asterisk fueron AGI para scripts y AMI para enviar comandos y recibir eventos a través de sockets. ARI representa la evolución hacia una interfaz más moderna y basada en REST para Asterisk.
This document summarizes Pascal Cadotte Michaud's presentation on getting a live transcript of calls from Asterisk. It discusses three main ways to stream audio out of Asterisk: RES_ARI_STREAM, Asterisk's external media functionality, and AudioSocket. RES_ARI_STREAM uses the Asterisk ARI WebSocket to stream audio. External media allows Asterisk to send RTP to an external host and port. AudioSocket is a simple TCP-based protocol for streaming audio. The document provides code examples and links to demonstrate each method. It encourages connecting with the Wazo Platform team to learn more about building solutions using open source telecom components.
This document discusses how to configure Kamailio as a session border controller (SBC) using Docker. It provides an introduction to Kamailio and explains how to install Kamailio from source on a CentOS Docker container. Specific steps are outlined to configure Kamailio to use MySQL, create the database schema, modify the configuration file to use the private and public IP addresses, and start the Kamailio server. The document also provides instructions for testing Kamailio by getting the error log, creating SIP user accounts, and registering a SIP softphone.
This document summarizes a presentation about Kamailio. Kamailio is an open source SIP server that can function as a proxy, registrar, location server, application server, dispatcher and websocket server. It has a modular architecture that allows additional functionality through modules. The presentation covered what's new in Kamailio 5.4 like new modules for Kafka, MQTT, dialog tracking and STIR/SHAKEN, as well as common deployment scenarios and examples of Kamailio configurations.
Kamailio combined with Asterisk creates and incredibly robust and durable VoIP framework. With scalability and security, adding Kamailio to an asterisk deployment makes sense and saves money.
This document provides an overview and objectives for an E-Learning training course on building a complete PBX with Asterisk. The training will cover understanding and installing Asterisk, building a simple PBX with SIP phones and SIP trunks, configuring features like dialplans, IVRs and voicemail, and integrating applications. By the end of the course, students will have hands-on experience building their own basic Asterisk system. Various chapters include objectives, presentations on topics, and instructions for labs to gain practical experience configuring Asterisk.
Using Asterisk and Kamailio for Reliable, Scalable and Secure Communication S...Fred Posner
Presentation from AsteriskWorld 2017 at ITEXPO. Discussion of how I started with Asterisk and Kamailio as well as how to build Reliability, Scalability, and Security into your telephony platform.
Using Kamailio for Scalability and SecurityFred Posner
Fred Posner discusses using Kamailio, an open source SIP server, for scalability and security. Kamailio can handle thousands of call setups per second through its modular design. It supports features like load balancing, TLS, filtering, rate limiting, and topology hiding to improve security and scalability. Fred highlights how Kamailio's flexibility through modules and programming interfaces allows customizing it for various deployment needs.
SIP Attack Handling (Kamailio World 2021)Fred Posner
This document discusses SIP attack handling in Kamailio. It provides an overview of common SIP attacks like denial of service and examples of attack traffic. It then outlines several Kamailio modules that can help mitigate attacks, including PIKE for rate limiting, SECFILTER for blocking lists, and HTABLE for temporary blocking. The document also discusses Kamailio configuration options for attack handling using these modules as well as other techniques like Fail2Ban, iptables-api, and APIBAN for sharing block lists.
LF_APIStrat17_Creating Communication Applications using the Asterisk RESTFul ...LF_APIStrat
The document provides an overview of the Asterisk RESTful Interface (ARI) and demonstrates how to build communication applications using ARI. It discusses the history and purpose of ARI and compares it to other Asterisk APIs like AGI and AMI. The document then demonstrates connecting to ARI via a WebSocket connection to receive events and making REST requests to play audio on a channel. It provides examples of the resources and categories available in the ARI specification.
Presentation from Asterisk conference on designing a High Availability Asterisk cluster. Covers the current state of HA, available open source products vs closed source, VoIP service continuity, synchronization of configuration, etc.
Presentation done at Kamailio World 2013, Berlin, Germany - several options for scalability of SIP routing with Kamailio, from configuration file tricks to stateless and stateful load balancing with dispatcher module.
Performance comparison between different versions of Asterisk and different channels.
Presentation at Astricon 2017
Online Trainings
Complete Asterisk PBX Training Coupon http://bit.ly/2E6U7fP
Understanding and Troubleshooting SIP http://bit.ly/2WZKkzy
VoIP Design Training Coupon http://bit.ly/2tlLFmG
VoIP Hacking Training Coupon http://bit.ly/2X4Pjz7
Quick Start to OpenSIPS Coupon http://bit.ly/2Bt72XJ
Books
Complete Asterisk Training Paperback and Kindle Book https://amzn.to/2tm7TFb
Complete Asterisk Training eBook PDF http://bit.ly/2UUebHG
Building Telephony Systems with OpenSIPS http://bit.ly/2SsUt9a
Treinamento Online
SIP em Profundidade Coupon de Treinamento http://bit.ly/2GpatTq
Asterisk Essencial Coupon de Treinamento http://bit.ly/2BzCjs1
Livros:
Guia e Configuração do Asterisk Paperback e Kindle https://amzn.to/2S2lwmZ
Guia de Configuração do Asterisk eBook PDF http://www.asteriskguide.com
High Availability Asterisk and FreePBX on Microsoft AzureSanjay Willie
In this slide, we presented to MaGIC Malaysia for entrepreneurs wanting to get an Asterisk business on cloud going. Here, we provide the most basic, lowest level method of having a HA on Microsoft Azure with FreePBX,
Components used with Azure's Ubuntu 14.04 image
1) Asterisk 11
2) FreePBX 2.11
3) DRBD
4) Heartbeat - but without actually relying on Virtual IP since we use Azure Cloud to do the HA by exposing TCP5060
5) Use Azure's HA method to achieve HA by ensuring that whichever server that "listens" to port TCP5060 will have all the HA DNS name of Asterisk traffic redirected there.
Podman is an open source tool for managing OCI containers and container images. It allows users to find, run, build, share and deploy applications using containers. Some key points about Podman include:
- It is daemonless, secure, and designed for Linux containers.
- Podman manages the entire container lifecycle from creation to deletion. It handles mounting, networking, and the container runtime.
- When running a container, Podman generates an OCI specification, pulls the image if needed, configures networking using Netavark, and uses Conmon to monitor the container process.
- Podman 4 introduced a new network stack based on Netavark and Aardvark-dns
This document discusses configuring FreeSWITCH, an open source telephony platform, on Docker. It provides background on IP telephony systems and introduces FreeSWITCH and Docker. The benefits of using FreeSWITCH on Docker are explained, such as easier deployment and scalability. Steps are outlined to install FreeSWITCH from source on a CentOS Docker container and configure the network settings. Finally, instructions are given to configure a SIP phone like Linphone and verify the FreeSWITCH installation by placing test calls.
[KubeCon EU 2022] Running containerd and k3s on macOSAkihiro Suda
https://sched.co/ytpi
It has been very hard to use Mac for developing containerized apps. A typical way is to use Docker for Mac, but it is not FLOSS. Another option is to install Docker and/or Kubernetes into VirtualBox, often via minikube, but it doesn't propagate localhost ports, and VirtualBox also doesn't support the ARM architecture. This session will show how to run containerd and k3s on macOS, using Lima and Rancher Desktop. Lima wraps QEMU in a simple CLI, with neat features for container users, such as filesystem sharing and automatic localhost port forwarding, as well as DNS and proxy propagation for enterprise networks. Rancher Desktop wraps Lima with k3s integration and GUI.
This document discusses the evosip platform, which uses Docker and Kubernetes to provide a scalable VoIP infrastructure based on Kamailio, Asterisk, and RTPEngine. Key aspects include:
- Using containers and Kubernetes for fast, automatic scaling with no limits and distributed architecture.
- Implementing Kamailio, Asterisk, and RTPEngine as stateless services using techniques like cached dispatchers, authentication from a shared table, and storing dialogs in a database.
- Using macvlan networking to give containers direct public IPs without NAT for better performance.
- Separating data and core service networks and using Multus CNI to give containers multiple networks.
-
Three Ways Kamailio Can Help Your FreeSWITCH DeploymentFred Posner
Kamailio can help a FreeSWITCH deployment in three ways: 1) Using the DISPATCHER module for carrier and internal routing which provides load balancing and failover capabilities, 2) Using the PERMISSIONS module for IP-based access control lists for routing, registrations, and permissions, and 3) Using the HTABLE module for caching and storing data in-memory for improved performance and scalability. Kamailio offers high performance, scalability, and security features that can enhance an existing FreeSWITCH system.
Considerable improvements can be achieved by automating the integration of Kamailio-based projects: automated builds, tests and deployments save time and increase reliability. This presentation focuses on common practices to automate the build of Kamailio (and RTPEngine) on various distributions and deploy them, together with their configuration, on testing and production environments.
Docker plays an important role in providing flexible, clean building environments and keep the process reproducible. We’ll see how Jenkins can orchestrate the builds with Docker slaves, and perform the deployments with a combination of platform-specific packages, Fabric, Puppet and Ansible.
Asterisk WebRTC frontier: realize client SIP Phone with sipML5 and Janus GatewayAlessandro Polidori
This document discusses integrating WebRTC phone capabilities into a browser using sipML5 and Janus. sipML5 is an open-source HTML5 SIP client that uses WebRTC for audio and video calls without plugins. Janus is a general purpose WebRTC gateway that can be used with a SIP plugin to enable calls. The document outlines how sipML5 uses WebRTC protocols for signaling and media, and provides code examples for initialization and calls. It then explains how Janus acts as a gateway between a browser-based client and a PBX, and provides similar code examples. Overall, the document proposes using sipML5 for a browser phone with Janus as a gateway to enable calls to a NethServer VoIP PBX.
A presentation about combining the powers of the Go language and Asterisk in order to provide fast, reliable and hugely scalable voice applications. A brief introduction about why Go presents a big opportunity for the asterisk community, a primer into developing FastAGI applications, demonstration of the AGI package [1] and a walk through the idioms and challenges of developing in Go. The talk is focused mainly on our experience of porting existing code into Go with the aim to scale our services to larger numbers accompanied with benchmarks and an introduction to some tools [2] we developed to help us test, debug and benchmark AGI applications.
Fred Posner gives a presentation on Kamailio, an open source SIP proxy/registrar. He introduces Kamailio and discusses what it is, its main functions and modules. Posner explains why Kamailio is useful, highlighting its security features, scalability, ability to bridge connections, and high availability. He provides examples of how Kamailio can integrate with and offload work from FreeSWITCH to handle large numbers of users in a secure and scalable way.
This document discusses using Kubernetes to deploy a FreeSWITCH cluster on Minikube for local development and testing. It covers creating a FreeSWITCH deployment, exposing it publicly via a service, scaling the deployment to multiple instances, and updating the deployment with zero downtime using rolling updates. The goal is to explore deploying and managing a FreeSWITCH application on Kubernetes.
Asterisk is an Open Source PBX - but how does it support larger installations? Can you scale it up to thousands of users, with hundreds of simultaneous calls? What about failover, backups and the famous blinking lamps? Olle Johansson goes through various models and describes where some of his current projects with strange names - Pinefrog, Pinana, Pinetree and Bufo fits into this picture.
This document discusses using the Asterisk REST Interface (ARI) to bridge calls across multiple Asterisk instances in a scaled environment. It explains that ARI allows remote control of Asterisk instances, which is necessary to connect calls handled by different instances. The main challenges are identifying conference calls, checking for existing conferences, and adding calls to conferences across instances. It proposes using a global database, routing keys, and one controlling instance to orchestrate conferences among multiple Asterisk servers.
Using Kamailio for Scalability and SecurityFred Posner
Fred Posner discusses using Kamailio, an open source SIP server, for scalability and security. Kamailio can handle thousands of call setups per second through its modular design. It supports features like load balancing, TLS, filtering, rate limiting, and topology hiding to improve security and scalability. Fred highlights how Kamailio's flexibility through modules and programming interfaces allows customizing it for various deployment needs.
SIP Attack Handling (Kamailio World 2021)Fred Posner
This document discusses SIP attack handling in Kamailio. It provides an overview of common SIP attacks like denial of service and examples of attack traffic. It then outlines several Kamailio modules that can help mitigate attacks, including PIKE for rate limiting, SECFILTER for blocking lists, and HTABLE for temporary blocking. The document also discusses Kamailio configuration options for attack handling using these modules as well as other techniques like Fail2Ban, iptables-api, and APIBAN for sharing block lists.
LF_APIStrat17_Creating Communication Applications using the Asterisk RESTFul ...LF_APIStrat
The document provides an overview of the Asterisk RESTful Interface (ARI) and demonstrates how to build communication applications using ARI. It discusses the history and purpose of ARI and compares it to other Asterisk APIs like AGI and AMI. The document then demonstrates connecting to ARI via a WebSocket connection to receive events and making REST requests to play audio on a channel. It provides examples of the resources and categories available in the ARI specification.
Presentation from Asterisk conference on designing a High Availability Asterisk cluster. Covers the current state of HA, available open source products vs closed source, VoIP service continuity, synchronization of configuration, etc.
Presentation done at Kamailio World 2013, Berlin, Germany - several options for scalability of SIP routing with Kamailio, from configuration file tricks to stateless and stateful load balancing with dispatcher module.
Performance comparison between different versions of Asterisk and different channels.
Presentation at Astricon 2017
Online Trainings
Complete Asterisk PBX Training Coupon http://bit.ly/2E6U7fP
Understanding and Troubleshooting SIP http://bit.ly/2WZKkzy
VoIP Design Training Coupon http://bit.ly/2tlLFmG
VoIP Hacking Training Coupon http://bit.ly/2X4Pjz7
Quick Start to OpenSIPS Coupon http://bit.ly/2Bt72XJ
Books
Complete Asterisk Training Paperback and Kindle Book https://amzn.to/2tm7TFb
Complete Asterisk Training eBook PDF http://bit.ly/2UUebHG
Building Telephony Systems with OpenSIPS http://bit.ly/2SsUt9a
Treinamento Online
SIP em Profundidade Coupon de Treinamento http://bit.ly/2GpatTq
Asterisk Essencial Coupon de Treinamento http://bit.ly/2BzCjs1
Livros:
Guia e Configuração do Asterisk Paperback e Kindle https://amzn.to/2S2lwmZ
Guia de Configuração do Asterisk eBook PDF http://www.asteriskguide.com
High Availability Asterisk and FreePBX on Microsoft AzureSanjay Willie
In this slide, we presented to MaGIC Malaysia for entrepreneurs wanting to get an Asterisk business on cloud going. Here, we provide the most basic, lowest level method of having a HA on Microsoft Azure with FreePBX,
Components used with Azure's Ubuntu 14.04 image
1) Asterisk 11
2) FreePBX 2.11
3) DRBD
4) Heartbeat - but without actually relying on Virtual IP since we use Azure Cloud to do the HA by exposing TCP5060
5) Use Azure's HA method to achieve HA by ensuring that whichever server that "listens" to port TCP5060 will have all the HA DNS name of Asterisk traffic redirected there.
Podman is an open source tool for managing OCI containers and container images. It allows users to find, run, build, share and deploy applications using containers. Some key points about Podman include:
- It is daemonless, secure, and designed for Linux containers.
- Podman manages the entire container lifecycle from creation to deletion. It handles mounting, networking, and the container runtime.
- When running a container, Podman generates an OCI specification, pulls the image if needed, configures networking using Netavark, and uses Conmon to monitor the container process.
- Podman 4 introduced a new network stack based on Netavark and Aardvark-dns
This document discusses configuring FreeSWITCH, an open source telephony platform, on Docker. It provides background on IP telephony systems and introduces FreeSWITCH and Docker. The benefits of using FreeSWITCH on Docker are explained, such as easier deployment and scalability. Steps are outlined to install FreeSWITCH from source on a CentOS Docker container and configure the network settings. Finally, instructions are given to configure a SIP phone like Linphone and verify the FreeSWITCH installation by placing test calls.
[KubeCon EU 2022] Running containerd and k3s on macOSAkihiro Suda
https://sched.co/ytpi
It has been very hard to use Mac for developing containerized apps. A typical way is to use Docker for Mac, but it is not FLOSS. Another option is to install Docker and/or Kubernetes into VirtualBox, often via minikube, but it doesn't propagate localhost ports, and VirtualBox also doesn't support the ARM architecture. This session will show how to run containerd and k3s on macOS, using Lima and Rancher Desktop. Lima wraps QEMU in a simple CLI, with neat features for container users, such as filesystem sharing and automatic localhost port forwarding, as well as DNS and proxy propagation for enterprise networks. Rancher Desktop wraps Lima with k3s integration and GUI.
This document discusses the evosip platform, which uses Docker and Kubernetes to provide a scalable VoIP infrastructure based on Kamailio, Asterisk, and RTPEngine. Key aspects include:
- Using containers and Kubernetes for fast, automatic scaling with no limits and distributed architecture.
- Implementing Kamailio, Asterisk, and RTPEngine as stateless services using techniques like cached dispatchers, authentication from a shared table, and storing dialogs in a database.
- Using macvlan networking to give containers direct public IPs without NAT for better performance.
- Separating data and core service networks and using Multus CNI to give containers multiple networks.
-
Three Ways Kamailio Can Help Your FreeSWITCH DeploymentFred Posner
Kamailio can help a FreeSWITCH deployment in three ways: 1) Using the DISPATCHER module for carrier and internal routing which provides load balancing and failover capabilities, 2) Using the PERMISSIONS module for IP-based access control lists for routing, registrations, and permissions, and 3) Using the HTABLE module for caching and storing data in-memory for improved performance and scalability. Kamailio offers high performance, scalability, and security features that can enhance an existing FreeSWITCH system.
Considerable improvements can be achieved by automating the integration of Kamailio-based projects: automated builds, tests and deployments save time and increase reliability. This presentation focuses on common practices to automate the build of Kamailio (and RTPEngine) on various distributions and deploy them, together with their configuration, on testing and production environments.
Docker plays an important role in providing flexible, clean building environments and keep the process reproducible. We’ll see how Jenkins can orchestrate the builds with Docker slaves, and perform the deployments with a combination of platform-specific packages, Fabric, Puppet and Ansible.
Asterisk WebRTC frontier: realize client SIP Phone with sipML5 and Janus GatewayAlessandro Polidori
This document discusses integrating WebRTC phone capabilities into a browser using sipML5 and Janus. sipML5 is an open-source HTML5 SIP client that uses WebRTC for audio and video calls without plugins. Janus is a general purpose WebRTC gateway that can be used with a SIP plugin to enable calls. The document outlines how sipML5 uses WebRTC protocols for signaling and media, and provides code examples for initialization and calls. It then explains how Janus acts as a gateway between a browser-based client and a PBX, and provides similar code examples. Overall, the document proposes using sipML5 for a browser phone with Janus as a gateway to enable calls to a NethServer VoIP PBX.
A presentation about combining the powers of the Go language and Asterisk in order to provide fast, reliable and hugely scalable voice applications. A brief introduction about why Go presents a big opportunity for the asterisk community, a primer into developing FastAGI applications, demonstration of the AGI package [1] and a walk through the idioms and challenges of developing in Go. The talk is focused mainly on our experience of porting existing code into Go with the aim to scale our services to larger numbers accompanied with benchmarks and an introduction to some tools [2] we developed to help us test, debug and benchmark AGI applications.
Fred Posner gives a presentation on Kamailio, an open source SIP proxy/registrar. He introduces Kamailio and discusses what it is, its main functions and modules. Posner explains why Kamailio is useful, highlighting its security features, scalability, ability to bridge connections, and high availability. He provides examples of how Kamailio can integrate with and offload work from FreeSWITCH to handle large numbers of users in a secure and scalable way.
This document discusses using Kubernetes to deploy a FreeSWITCH cluster on Minikube for local development and testing. It covers creating a FreeSWITCH deployment, exposing it publicly via a service, scaling the deployment to multiple instances, and updating the deployment with zero downtime using rolling updates. The goal is to explore deploying and managing a FreeSWITCH application on Kubernetes.
Asterisk is an Open Source PBX - but how does it support larger installations? Can you scale it up to thousands of users, with hundreds of simultaneous calls? What about failover, backups and the famous blinking lamps? Olle Johansson goes through various models and describes where some of his current projects with strange names - Pinefrog, Pinana, Pinetree and Bufo fits into this picture.
This document discusses using the Asterisk REST Interface (ARI) to bridge calls across multiple Asterisk instances in a scaled environment. It explains that ARI allows remote control of Asterisk instances, which is necessary to connect calls handled by different instances. The main challenges are identifying conference calls, checking for existing conferences, and adding calls to conferences across instances. It proposes using a global database, routing keys, and one controlling instance to orchestrate conferences among multiple Asterisk servers.
An Asterisk administrator discusses upgrading an Asterisk PBX from version 11 to version 16. This involved building new Debian packages, porting configurations, updating dialplans, testing features, and addressing issues like crashes, logging problems, and high system loads. Through testing and troubleshooting, the upgrade was eventually successful and the system could handle the same call load as the previous version. However, ongoing work is still needed to optimize logging and reduce continuous CPU loads.
SITREP - Asterisk REST. The first steps are done, now what? - CommCon 2019Jöran Vinzens
- The current Asterisk infrastructure is complex, outdated, and difficult to maintain. It uses custom patches and an old version of chan_sip.
- The team wants to update components, simplify interactions, and eliminate custom code to make the system more secure, scalable, and able to utilize new Asterisk features.
- They plan to use Asterisk REST Interface (ARI) to create a new interface and decouple Asterisk from the call controller for improved scalability. This will involve a step-by-step "soft" migration rather than a full rewrite.
This document discusses the motivation, design, and implementation of a distributed high availability Asterisk application using Stasis and ARI over Kafka. The current setup uses AGI and AMI with an evolved monolithic architecture. The new approach uses ARI, modularizes the system, and uses a single Asterisk server. Kamailio dispatches calls and observes Asterisk. A Stasis app handles SIP/media, and a call controller manages call logic and routing. ARI events are sent to Kafka, and commands are sent back. This allows for transparent server farms, easy scaling, and restart safety while meeting demands for high availability, performance, scalability, and continuous deployment.
The document discusses using Ansible to automate the deployment of VoIP components. It begins by describing the manual deployment process and then outlines the benefits of using Ansible including continuous deployment, automatic testing and rollback. It provides details on how Ansible works with Jenkins to deploy configuration changes to VoIP systems after code is pushed to GitHub. The document emphasizes that VoIP systems require special handling during deployment due to ongoing calls and real-time requirements. It suggests separating code and configuration and describes using Ansible to template configuration files and deploy them to different hosts, while handling things like reloading services.
sipgate built a carrier grade VoIP infrastructure in Germany over 10 years, starting as a small VoIP carrier in 2004 and expanding to interconnect with telecom carriers via SS7 and support carrier features. Their infrastructure now uses YATE to handle SS7 interconnect, MAP/CAMEL protocols and control media gateways, Asterisk for billing, announcements and PBX features, Kamailio as a SIP back-to-back user agent and stateful proxy, and monitoring tools to oversee the complex signaling and media routing across systems.
The document discusses using Homer to debug ISUP/SIP calls by capturing network traffic. Homer allows capturing SIP, ISUP, and other protocols in one tool. It involves installing a capture agent to collect traffic, sending it to a capture node like Kamailio that processes the data, and storing it in a MySQL database. A web interface called Webhomer provides a GUI to view analyzed calls and troubleshoot issues across different protocols in a single tool.
The document discusses securing VoIP systems against common fraud scenarios like international revenue share fraud, payment fraud, denial of service attacks, and identity theft. It provides tips for securing the network, SIP service, phones/endpoints, and administering the system through measures like firewalls, strong passwords, call limits, whitelisting valid numbers and destinations, monitoring, and being vigilant against fraud. The overarching message is that fraud is an ongoing issue requiring continuous effort to improve security.
Astricon 2017 Superpower of deployment toolsJöran Vinzens
The document discusses using deployment tools like Ansible to automate configuration management. Previously, manual deployment and patching occurred every two weeks. Ansible can enable continuous deployment, automatic testing and rollback, and remove the need for specialized knowledge. It works by triggering Jenkins when code is pushed, building packages, running Ansible scripts to install packages, configure systems, run tests, and rollback if needed. Templates allow host-specific configurations. Ansible can perform tasks across multiple systems and be used to automate common administration tasks.
11June 2024. An online pre-engagement session was organized on Tuesday June 11 to introduce the Science Policy Lab approach and the main components of the conceptual framework.
About 40 experts from around the globe gathered online for a pre-engagement session, paving the way for the first SASi-SPi Science Policy Lab event scheduled for June 18-19, 2024 in Malmö. The session presented the objectives for the upcoming Science Policy Lab (S-PoL), which featured a role-playing game designed to simulate stakeholder interactions and policy interventions for food systems transitions. Participants called for the sharing of meeting materials and continued collaboration, reflecting a strong commitment to advancing towards sustainable agrifood systems.
• For a full set of 530+ questions. Go to
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• SkillCertPro offers detailed explanations to each question which helps to understand the concepts better.
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This presentation by Professor Giuseppe Colangelo, Jean Monnet Professor of European Innovation Policy, was made during the discussion “The Intersection between Competition and Data Privacy” held at the 143rd meeting of the OECD Competition Committee on 13 June 2024. More papers and presentations on the topic can be found at oe.cd/ibcdp.
This presentation was uploaded with the author’s consent.
This presentation by Katharine Kemp, Associate Professor at the Faculty of Law & Justice at UNSW Sydney, was made during the discussion “The Intersection between Competition and Data Privacy” held at the 143rd meeting of the OECD Competition Committee on 13 June 2024. More papers and presentations on the topic can be found at oe.cd/ibcdp.
This presentation was uploaded with the author’s consent.
1.) Introduction
Our Movement is not new; it is the same as it was for Freedom, Justice, and Equality since we were labeled as slaves. However, this movement at its core must entail economics.
2.) Historical Context
This is the same movement because none of the previous movements, such as boycotts, were ever completed. For some, maybe, but for the most part, it’s just a place to keep your stable until you’re ready to assimilate them into your system. The rest of the crabs are left in the world’s worst parts, begging for scraps.
3.) Economic Empowerment
Our Movement aims to show that it is indeed possible for the less fortunate to establish their economic system. Everyone else – Caucasian, Asian, Mexican, Israeli, Jews, etc. – has their systems, and they all set up and usurp money from the less fortunate. So, the less fortunate buy from every one of them, yet none of them buy from the less fortunate. Moreover, the less fortunate really don’t have anything to sell.
4.) Collaboration with Organizations
Our Movement will demonstrate how organizations such as the National Association for the Advancement of Colored People, National Urban League, Black Lives Matter, and others can assist in creating a much more indestructible Black Wall Street.
5.) Vision for the Future
Our Movement will not settle for less than those who came before us and stopped before the rights were equal. The economy, jobs, healthcare, education, housing, incarceration – everything is unfair, and what isn’t is rigged for the less fortunate to fail, as evidenced in society.
6.) Call to Action
Our movement has started and implemented everything needed for the advancement of the economic system. There are positions for only those who understand the importance of this movement, as failure to address it will continue the degradation of the people deemed less fortunate.
No, this isn’t Noah’s Ark, nor am I a Prophet. I’m just a man who wrote a couple of books, created a magnificent website: http://www.thearkproject.llc, and who truly hopes to try and initiate a truly sustainable economic system for deprived people. We may not all have the same beliefs, but if our methods are tried, tested, and proven, we can come together and help others. My website: http://www.thearkproject.llc is very informative and considerably controversial. Please check it out, and if you are afraid, leave immediately; it’s no place for cowards. The last Prophet said: “Whoever among you sees an evil action, then let him change it with his hand [by taking action]; if he cannot, then with his tongue [by speaking out]; and if he cannot, then, with his heart – and that is the weakest of faith.” [Sahih Muslim] If we all, or even some of us, did this, there would be significant change. We are able to witness it on small and grand scales, for example, from climate control to business partnerships. I encourage, invite, and challenge you all to support me by visiting my website.
Gamify it until you make it Improving Agile Development and Operations with ...Ben Linders
So many challenges, so little time. While we’re busy developing software and keeping it operational, we also need to sharpen the saw, but how? Gamification can be a way to look at how you’re doing and find out where to improve. It’s a great way to have everyone involved and get the best out of people.
In this presentation, Ben Linders will show how playing games with the DevOps coaching cards can help to explore your current development and deployment (DevOps) practices and decide as a team what to improve or experiment with.
The games that we play are based on an engagement model. Instead of imposing change, the games enable people to pull in ideas for change and apply those in a way that best suits their collective needs.
By playing games, you can learn from each other. Teams can use games, exercises, and coaching cards to discuss values, principles, and practices, and share their experiences and learnings.
Different game formats can be used to share experiences on DevOps principles and practices and explore how they can be applied effectively. This presentation provides an overview of playing formats and will inspire you to come up with your own formats.
This presentation by Tim Capel, Director of the UK Information Commissioner’s Office Legal Service, was made during the discussion “The Intersection between Competition and Data Privacy” held at the 143rd meeting of the OECD Competition Committee on 13 June 2024. More papers and presentations on the topic can be found at oe.cd/ibcdp.
This presentation was uploaded with the author’s consent.
8. Using ARI and AGI 8
Basics
Start at the beginning
PBX
Alice
Bob
Charli
e
Ed David
Trunk Carrier
9. Using ARI and AGI 9
Start at the beginning
Basics
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
Kamailio
Kamailio
Carrier
10. Using ARI and AGI 10
Start at the beginning
Basics
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
Kamailio
Kamailio
Carrier
11. Using ARI and AGI 11
Basics
Start at the beginning
Dialplan
extensions.conf
12. Using ARI and AGI 12
Basics
Asterisk dialplan
exten = 10,1,Log(INFO, Hello World)
13. Using ARI and AGI 13
Basics
Asterisk dialplan
exten = 10,1,Log(INFO, Hello World)
extension
priority
application
parameter
14. Using ARI and AGI 14
Basics
Asterisk dialplan
exten = 10,1,Log(INFO, Hello World)
same = n,Answer()
same = n,Playback(tt-monkeys)
same = n,Hangup()
15. Using ARI and AGI 15
Basics
Asterisk dialplan
exten = _XN.,1,Dial(PJSIP/${EXTEN}@trunk-out,120tr)
16. Using ARI and AGI 16
Basics
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_Dial
17. Using ARI and AGI 17
Basics
Asterisk dialplan
there are never enough features
18. Using ARI and AGI 18
Basics
Asterisk dialplan
AGI
ARI
AMI
19. Using ARI and AGI 19
AGI vs ARI
Asterisk dialplan
AGI
Asterisk Gateway Interface
ARI
Asterisk REST Interface
21. Using ARI and AGI 21
AGI vs ARI
AGI / FastAGI
Trigger external script
Query external service
22. Using ARI and AGI 22
AGI vs ARI
AGI
"AGI is analogous to CGI in Apache. AGI provides an interface between the Asterisk dialplan and an external
program that wants to manipulate a channel in the dialplan. In general, the interface is synchronous - actions
taken on a channel from an AGI block and do not return until the action is completed."
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=32375589
24. Using ARI and AGI 24
AGI vs ARI
AGI use cases
DB lookups
LCR lookup
External accounting
25. Using ARI and AGI 25
AGI vs ARI
Asterisk dialplan
exten = _XN.,1,AGI(someLogic.agi)
exten = _XN.,1,AGI(agi://1.2.3.4:1234/entrypoint)
26. Using ARI and AGI 26
AGI vs ARI
AGI
● Dialplan application
● Only when dialplan is processed
● Blocks Dialplan
● No control during call
27. Using ARI and AGI 27
AGI vs ARI
ARI
Websocket / REST Interface
28. Using ARI and AGI 28
AGI vs ARI
ARI
"ARI is an asynchronous API that allows developers to build communications applications by exposing the raw primitive objects in
Asterisk - channels, bridges, endpoints, media, etc. - through an intuitive REST interface."
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
29. Using ARI and AGI 29
AGI vs ARI
Asterisk dialplan
exten = _XN.,1,Stasis(myStasisApp)
30. Using ARI and AGI 30
AGI vs ARI
JSON
{
"application":"hello-world",
"type":"StasisStart",
"timestamp":"2014-05-20T13:15:27.131-0500",
"args":[],
"channel":{
"id":"1400609726.3",
"state":"Up",
"name":"PJSIP/1000-00000001",
"caller":{
"name":"",
"number":""},
"connected":{
"name":"",
"number":""},
"accountcode":"",
"dialplan":{
"context":"default",
"exten":"1000",
"priority":1},
"creationtime":"2014-05-20T13:15:26.628-0500"}
}
31. Using ARI and AGI 31
AGI vs ARI
ARI
● Basic building blocks
● Blocks Dialplan
● JSON
32. Using ARI and AGI 32
AGI vs ARI
ARI building blocks
● Channels
● Bridges
● Playbacks
● Recordings
● ...
33. Using ARI and AGI 33
AGI vs ARI
ARI building blocks
Channel Channel
Bridge
34. Using ARI and AGI 34
ARI building blocks
AGI vs ARI
Channel Channel
Bridge
Asterisk
Alice Bob
SIP Invite - PJSIP Channel POST ari/channels/create - PJSIP Channel
35. Using ARI and AGI 35
AGI vs ARI
ARI
1. PJSIP channel incoming Invite
2. Dialplan App Stasis() -> StasisStart
3. REST POST bridge create
4. REST POST channel create
5. REST POST bridge add both channels
6. REST POST channel dial
7. PJSIP channel outgoing Invite
36. Using ARI and AGI 36
AGI vs ARI
ARI
● More work than dialplan
● More components than asterisk only
● Yet another API to learn
● ...
Why should one do that?
38. Using ARI and AGI 38
How to interact
AGI and ARI
Why use both?
39. Using ARI and AGI 39
How to interact
AGI and ARI
e.g.
select Stasis App in AGI
40. Using ARI and AGI 40
How to interact
AGI and ARI
exten = _XN.,1,Agi(selectStasisApp.agi)
same = n,Log(NOTICE, AGI selected ARI App: ${STASISAPP})
same = n,Stasis(${STASISAPP})
41. Using ARI and AGI 41
ARI primitive objects
AGI vs ARI
Dialplan Channel
Asterisk
Alice Bob
SIP Invite - PJSIP Channel POST ari/channels/create - PJSIP Channel
AGI ARI
Bridge
42. Using ARI and AGI 42
How to interact
AGI and ARI
What if we want to call an AGI on a ARI
initiated call?
43. Using ARI and AGI 43
ARI primitive objects
AGI vs ARI
Channel -> Dialplan Channel
Asterisk
Alice Bob
SIP Invite - PJSIP Channel POST ari/channels/create - PJSIP Channel
AGI ARI
Bridge
AGI
44. Using ARI and AGI 44
How to interact
AGI and ARI
The magic of Local Channel
45. Using ARI and AGI 45
How to interact
What is a local Channel?
Channel Channel
app_dial
Asterisk
Alice Bob
PJSIP Channel PJSIP Channel
46. Using ARI and AGI 46
How to interact
What is a local Channel?
Channel Channel
app_dial
Asterisk
Alice Bob
PJSIP Channel PJSIP Channel
app_dial
Channel
Local Channel
47. Using ARI and AGI 47
How to interact
How to use local Channels with Stasis?
Channel Channel
app_dial
Asterisk
Alice Bob
PJSIP Channel PJSIP Channel
app_dial
Channel
Local Channel
48. Using ARI and AGI 48
How to interact
How to use local Channels with Stasis?
Channel Channel
app_stasis
Asterisk
Alice Bob
PJSIP Channel PJSIP Channel
app_dial
Channel
Local Channel
49. Using ARI and AGI 49
How to interact
How to use local Channels with Stasis?
Channel Channel
app_stasis
Asterisk
Alice Bob
PJSIP Channel PJSIP Channel
app_dial
Channel
Local Channel
AGI AGI
50. Using ARI and AGI 50
How to interact
AGI and ARI
Alice AGI
Stasis
Dial
AGI
determine Stasis App
control over the entire call
place Local Channel
Check Wholesale Carrier (LCR)
place call
Carrier
51. Using ARI and AGI 51
Call handling with multiple Instances
Connect Asterisk instances?
52. Using ARI and AGI 52
Call handling with multiple Instances
Connect Asterisk instances?
First, a scenario
53. Using ARI and AGI 53
Call handling with multiple Instances
Connect Asterisk instances?
First, a scenario
Pickup a call!
54. Using ARI and AGI 54
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
55. Using ARI and AGI 55
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
56. Using ARI and AGI 56
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
57. Using ARI and AGI 57
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
AGI
What Stasis App?
58. Using ARI and AGI 58
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
AGI
What Stasis App?
59. Using ARI and AGI 59
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
AGI
What Stasis App?
60. Using ARI and AGI 60
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
AGI
What Stasis App?
61. Using ARI and AGI 61
Call handling with multiple Instances
Drawbacks
● Complex call routing
● Nothing for free from Asterisk
● Yet another API
● Shared knowledge over instances
62. Using ARI and AGI 62
Call handling with multiple Instances
Benefits
● Scaling
● Reduce fallout on crash
● REST / Websocket
● Enforced boundaries
63. Using ARI and AGI 63
Call handling with multiple Instances
Drawbacks
Single point of Failure
64. Using ARI and AGI 64
Call handling with multiple Instances
Kamailio
Alice
Bob
Charlie
Asterisk
Asterisk
Asterisk
Asterisk
Carrier
Kamailio
AGI
What Stasis App?
ARI
Call control
AGI
What Stasis App?
65. Using ARI and AGI 65
Call handling with multiple Instances
Asterisk
Asterisk
Asterisk
Asterisk
ARI
ARI
ARI
ARI
66. Using ARI and AGI 66
Call handling with multiple Instances
Asterisk
Asterisk
Asterisk
Asterisk
ARI
ARI
Use an ARI Proxy
Kafka / RabbitMQ / NATS
ARI
Proxy
ARI
Proxy
ARI
Proxy
ARI
Proxy
67. Using ARI and AGI 67
Call handling with multiple Instances
Asterisk
Asterisk
Asterisk
Asterisk
ARI
ARI
Use an ARI Proxy
Kafka / RabbitMQ / NATS
ARI
Proxy
ARI
Proxy
ARI
Proxy
ARI
Proxy
Control callflow ARI connection
68. Using ARI and AGI 68
Call handling with multiple Instances
ARI Proxy
CyCoreSystems
https://github.com/CyCoreSystems/ari-proxy - NATS / RabbitMQ
retel-io
https://github.com/retel-io/ari-proxy - Kafka
(Kafka sample https://github.com/vinzens81/ari-controller)
69. Using ARI and AGI 69
Call handling with multiple Instances
Nice picture here!