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

[Anton VG]

Kai,

could you please share your patch with us and describe more in detail what
does it do?

2006/3/15, Kai Militzer <km at westend.com>:
>
> 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
>
> --
> 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
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20060315/d96c0420/attachment-0001.htm