-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 11/02/2013 12:40, giovanni.v a ←crit : > On 11/02/2013 17.01, Jean-Denis Girard wrote: > I believe the first one will be not a viable option at all, no telco > will change any important protocol compliance rule on a "per subscriber" > basis.
Well, in this case the subscriber is also the telco! > Now forget your gateway for a moment and make a call on an imaginary > phone connected to your PRI, after that call successfully answered let > the called party hang up before you do. What you expect to hear? Sure, > a disconnect tone... so you will put your phone on hook and the phone > will send a disconnect immediately. > > Your pri<->gateway<->asterisk should work the same, even if the gateway > does not send a disconnect immediately the user who started that call > will hang up at least when hearing the disconnect tone (good feedback > for humans, no?) and asterisk will send a bye to the sip gateway then > that one shall initiate a disconnect on the user side. You're right, when someone answers it's not a problem. But if the caller is sent to voicemail and he hangups, we get 30 seconds disconnect tone. Or if the caller is sent to a queue and he hangs up, the agent takes a call which is already hung up. > Check also if your gateway allow for customization to remap isdn/q.931 > messages to sip. Yes, I already looked for such a configuration option, but unfortunately couldn't find any. I hoped someone on the list had experience with a Mediatrix on Euro ISDN. > Sorry, hope you will be able to understand because English isn't my > native language. No problem understanding, maybe because English is not my native language either ;) Thanks a lot, - -- Jean-Denis Girard SysNux Syst│mes Linux en Polyn←sie franaise http://www.sysnux.pf/ T←l: +689 50 10 40 / GSM: +689 79 75 27 -----BEGIN PGP SIGNATURE----- iEYEARECAAYFAlEacSkACgkQuu7Rv+oOo/g+cgCeMPAPVplRp2o/QvxnWGdoux5q BTwAnArY230E3pPU8pJ4/OprCUmax8Gs =kWdl -----END PGP SIGNATURE----- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users