[asterisk-ss7] My experiences with chan_ss7, some questions and a solution for the ringback tone

Kai Militzer km at westend.com
Wed Mar 15 08:07:30 MST 2006

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 

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 Militzer                 WESTEND GmbH  |  Internet-Business-Provider
Technik                      CISCO Systems Partner - Authorized Reseller
                              Lütticher Straße 10      Tel 0241/701333-14
km at westend.com               D-52064 Aachen              Fax 0241/911879

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