NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration

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NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration

Giovanni Maruzzelli
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
[hidden email]
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Re: NEW tutorial on Realtime OpenSIPS - FreeSWITCH Integration

shaheryarkh
Great stuff Maruzzelli. Thanks for sharing.

Thank you.


On Mon, Mar 4, 2013 at 7:05 PM, Giovanni Maruzzelli <[hidden email]> 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
[hidden email]
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: [hidden email]
Email: [hidden email]

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users