[Freeswitch-users] [OpenSIPS-Users] NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration

Muhammad Shahzad shaheryarkh at gmail.com
Tue Mar 5 00:38:32 MSK 2013


Great stuff Maruzzelli. Thanks for sharing.

Thank you.


On Mon, Mar 4, 2013 at 7:05 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> 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
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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