SS7 StackSistema de Señalización 7 (SS7) es un sistema de señalización que permite enviar más información útil para sistemas telefónicos de consumo, utilizado habitualmente por proveedores de telefonía y de servicios IP.

Esta mañana, nuestro compañero Matthew Fredrickson (Digium) ha terminado una serie de modificaciones en la librería libss7 que permitiría a Asterisk manejar de manera bastante más estable este tipo de señalización tan importante para determinadas empresas e instituciones y que estan incluidas en la versión subversion “Trunk” de Asterisk y libss7 que se descarga desde aquí:
http://svn.digium.com/view/

Por otro lado y en paralelo, en voip-info leía que Openvox anunciaba una serie de parches para estas librerías que no entiendo bien la relación con el apartado anterior emitido casi en el mismo instante. (ya lo comenta también voipnovatos)

Matthew envía a VentureVoIP (el nuevo nombre de Sineapps) un tutorial sobre cómo instalar la nueva versión y comprobar que funciona.

Tutorial de SS7: http://www.venturevoip.com/news.php?rssid=1843

To Use:
=======
In order to use libss7, you must get at least the following versions of
Zaptel, libss7, and Asterisk (you might also want to get libpri):
Zaptel: 1.4.x
Asterisk: trunk
libss7: trunk (currently, there *only* is a trunk release).

You must then do a `make; make install` in each of the directories that you installed in the given order (Zaptel first, libss7 second, and Asterisk last).

NOTE: In order to check out the code, you must have the subversion client installed. This is how to check them out from the public subversion server.

These are the commands you would type to install them:

`svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4`
`cd zaptel-1.4`
`make; make install`

`svn co http://svn.digium.com/svn/libss7/trunk libss7-trunk`
`cd libss7-trunk`
`make; make install`

`svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk`
`cd asterisk-trunk`
`./configure; make; make install;`

This should build Zaptel, libss7, and Asterisk with SS7 support.

In the past, there was a special asterisk-ss7 branch to use which contained the SS7 code. That code has been merged back into the trunk version of Asterisk, and the old asterisk-ss7 branch has been deprecated and removed. If you are still using the asterisk-ss7 branch, it will not work against the current version of libss7, and you should switch to asterisk-trunk instead.

CONFIGURATION:
In zaptel.conf, your signalling channel(s) should be a “dchan” and your bearers should be set as “bchan”.

In the asterisk-ss7 branch, there is a sample zapata.conf that is installed which contains sample configuration for setting up an E1 link.

In brief, here is a simple ss7 linkset setup:

signalling = ss7
ss7type = itu ; or ansi if you are using an ANSI link

linkset = 1 ; Pick a number for your linkset identifier in
; zapata.conf

pointcode = 28 ; The decimal form of your point code. If you
; are using an
; ANSI linkset, you can use the xxx-xxx-xxx
notation for
; specifying your linkset pointcode.
adjpointcode = 2 ; The point code of the switch adjacent to your
; linkset

defaultdpc = 3 ; The point code of the switch you want to send
; your ISUP
; traffic to. A lot of the time, this is the
; same as your
; adjpointcode.

; Now we configure our Bearer channels (CICs)

cicbeginswith = 1 ; Number to start counting the CICs from. So if
; Zap/1 to Zap/15 are CICs 1-15, you would set
; this to 1 before you
; declare channel=1-15

channel=1-15 ; Use Zap/1-15 and assign them to CICs 1-15

cicbeginswith = 17 ; Now for Zap/17 to Zap/31, they are CICs 17-31
; so we initialize
; cicbeginswith to 17 before we declare those
; channels

channel = 17-31 ; This assigns CICs 17-31 to channels 17-31

signchan = 16 ; This is where you declare which Zap channel is
; your signalling
; channel. In our case it is Zap/16. You can
; add redundant
; signalling channels by adding additional
; sigchan= lines.

; If we want an alternate redundant signalling channel add this

sigchan = 48 ; This would put two signalling channels
in our ; linkset, one at Zap/16 and one at Zap/48 which
; both would be used to send/receive
; ISUP traffic.

; End of zapata.conf

This is how a basic linkset is setup. For more detailed zapata.conf SS7 config information as well as other options available for that file, see the default zapata.conf that comes with the samples in asterisk. If you would like, you can do a `make samples` in your asterisk-trunk directory and it will install a sample zapata.conf for you that contains more information about SS7 setup.


Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

1 Comentario

  • Hola amigo,

    Tengo Elastix 1.3 instalado y compilé asterisk 1.6 para tener soporte para SS7 pero me daña la integración con Elastix y Freepbx. Me gustaría compilar el zaptel y el asterisk proporcionado por Elastix y el libss7 que colocas en tu articulo, ahora bien, mi preguntas son las siguientes: ¿El asterisk trunk es el 1.6? ¿puedo compilar el zaptel de elastix, el libss7 de tu articulo y el asterisk de elastix?

    Cualquier sugerencia te agradecería enormemente

    Gracias y Saludos

    Ronald García

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