Hey all! For the past year I have been working on and off on an SS7
implementation here at Digium called libss7. I have it to the point
where it can pass phone calls, so I figured it would be a good time to
release it and let people begin testing it. It's still somewhat "bare
bones" in functionality, but I've been doing a lot of fleshing out of
the implementation.
Currently, it has been used (making and receiving phone calls) and
developed in an ITU SS7 environment, but I have a good chunk of the
code included which is required for ANSI support as well. I think I'm
going to get an ANSI link in a few weeks, so hopefully I'll have that
tested and working relatively soon.
It supports MTP2, MTP3, and ISUP. After I get these layers fleshed
out, I'm planning on starting on SCCP and the layers above that with
the eventual goal of database-lookup and SMS support.
To test, you must have a T1/E1 card as well as an SS7 link. You also
need to have zaptel installed on your system.
Here are the instructions for checking it out of subversion and getting
it working:
`svn co http://svn.digium.com/svn/libss7/trunk libss7`
`cd libss7`
`make install`
Right now, the changes to chan_zap are implemented in a special
developer branch of asterisk. These are the instructions to check it
out
`svn co http://svn.digium.com/svn/asterisk/team/mattf/asterisk-ss7
asterisk-ss7`
`cd asterisk-ss7`
If you haven't compiled trunk yet, you may have to run `make` a few
times so that the configure script runs and sets things up properly.
It should find libss7, and compile chan_zap with support for it. The
link is brought up automatically when Asterisk starts.
Configuration in zaptel.conf is similar to that of a PRI. Your
signalling channel will be set as a "dchan" and the bearer channels are
set as "bchan". For information about setting up zapata.conf, see the
sample zapata.conf in the configs/zapata.conf.sample in the
asterisk-ss7 branch. There also is a libss7 project section on Mantis
now for any bugs that you might encounter.
Matthew Fredrickson
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