Hello comunity,
I tought I could share my experience with chan_ss7 with you and maybe
get some answers/opinions from the rest of you. How wants to know the
solution for the ringback tone will have to read til the end of this
mail. ;)
What is most important to know for the most, is I guess, that chan_ss7
works with an Alcatel S12 switch. I have it (in a lab config) running
relativly stable since late december 2005 (starting with chan_ss7-0.2)
with one E1 (30 Channels). Three weeks ago (befor I went on vacation ;)
) I added another E1 and this also seems to work (say: I was still able
to make calls after my return today).
The version I am currently running is modified version 0.8. The
modifications were neccesarry because I use chan_ss7 to "convert" calls
from ss7 to SIP and vice versa without terminating them on this asterisk
instance. The SIP part of the call is simply forwarded to a SIP Server
that then terminates the call. The problem I had was, that I cannot tell
on the asterisk with chan_ss7 if the dialed number is complete and
equiped and so I have to match everything with _X. This approach did not
work with overlap dialing, because it would match directly after the
IAM. So I added a timer that waits for a SAM after an IAM and starts
again if a SAM is received. In my opinion this is the only way to use
chan_ss7 as a gateway without knowledge of the numberingplan on the
final destination. Sifira didn't see it this way and wouldn't take my
patch into the main chan_ss7 ;( , maybe some of you could convince them
to do so. ;)
Another problem I had was with the handling of the hangupcause which
weren't translated correctly from SS7 to SIP and other way round. In my
opinion the error was in ast_softhangup_nolock in asterisk, but seems
not to be the case (see http://bugs.digium.com/view.php?id=6550).
@sifira, if you are reading this: would it be possible to fix this in
chan_ss7?
Now my question to the comunity: Is anyone running asterisk with
chan_ss7 as PSTN-to-SIP Gateway anf if yes what are your experiences?
Does it work reliable, what call volumes do you handle with it?
And last but not least, I also had the problem that no ringback tones
were generated by asterisk. The following two lines in the dialplan
inserted before the Dial statement do the trick:
exten => _X.,n,SetLanguage(de)
exten => _X.,n,Playtones(ring)
I hope that helps. ;)
Best regards,
Kai
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