Kai, could you please share your patch with us and describe more in detail what does it do? 2006/3/15, Kai Militzer <km@xxxxxxxxxxx>: > > 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 > L?tticher Stra?e 10 Tel 0241/701333-14 > km@xxxxxxxxxxx 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
|