On Wed, 15 Mar 2006, Kai Militzer wrote: > I tought I could share my experience with chan_ss7 with you and maybe > get some answers/opinions from the rest of you. > 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. ;) We didn't like the timer-solution because we think its wrong to delay all calls X seconds just because the SS7-asterisk doesn't know another Asterisk's dialplan. My suggestions is 1. Use identical dialplans on the SS7-gateway and the SIP server 2. Store the dialplan in a shared database. 3. I think it is (maybe) posible to 'share' the dialplan through IAX (anybody ?) > 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? We well look into this. Thank you for noticing. > 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? We at Sifira are also very interested in hearing any experiences :) > 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) We actually tried this, but we had to insert a ,1,Answer before the Playtones command. ...but the Answer before Playtones, breaks most telcos billing system, since a call is 'from the Answer to a hangup'. Anyway, we are working at a solution which will get chan_ss7 & Asterisk to generate indication-tones before the ANM has been sent to the remote switch. Mvh. Jacob -- Jacob Tinning System Developer SIFIRA A/S
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