Ciao VOIPers, it's my pleasure to bring to your attention a new tutorial on realtime integration between OpenSIPS and FreeSWITCH. It's a cut and paste tutorial, so you can test it right away, eg on a virtual machine, and when confident customize it and put it in production. The stack is Debian Squeeze 6.x, OpenSIPS 1.8.x, FreeSWITCH 1.2.x, OpenSIPS-CP as GUI, MySQL as database. You can find the tutorial at URL: http://www.opensips.org/Resources/DocsTutFreeSwitch with all required files. Please let us know what do you think about it, and what other tutorials you would like to read (at the moment I'm thinking at an HA install of FusionPBX+FreeSWITCH+OpenSIPS, but other requests will be taken into account too). See below for a small excerpt of this tutorial: ===== 1.1 Scope This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. This tutorial presents the concept and implementation of a realtime integration of OpenSIPS SIP server and FreeSWITCH media server. OpenSIPS is used a SIP server - users are registering with it, it routes calls, etc - while the purpose of FreeSWITCH is to provide a full set of media services - like voicemail, conference, announcements, etc. It is a realtime integration because both OpenSIPS and FreeSWITCH are provisioned in the same time when comes to user accounts - when creating a new OpenSIPS user, automatically FreeSWITCH will learn about it an provide and configure all necessary media services for it. Both OpenSIPS and FreeSWITCH will be provisioned (for user accounts) via a shared mysql database. All FreeSWITCH functionalities will be available to OpenSIPS users by prefixing "*" (eg: star) to the extension dialed. *1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234 ________________________________ 1.2 Setup presentation This tutorial can be used as a cut and paste complete and working installation. Please follow strictly all the steps, in the order given. The following services will be offered by FreeSWITCH by this integrated configuration: voicemail - users will get access to their mailbox; authentication will be done by OpenSIPS; while FreeSWITCH will only provide voicemail IVR (with access PIN); conference' - OpenSIPS will detect and forward calls related to conference service (based on prefixes) to FreeSWITCH, which will provide access (pin based) to the conference rooms; all functionalities - OpenSIPS users will prefix * to reach the corresponding extension in FreeSWITCH (*1234 will be passed to FreeSWITCH as 1234, while **1234 will be passed to FreeSWITCH as *1234) ===== ciao for now, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618