On Fri, Jun 06, 2014 at 08:28:07AM -0400, John Novack SCII wrote: > A J Stiles wrote: > >On Thursday 05 Jun 2014, Mojtaba wrote: > >>My scenario is (2) > >After doing some tests with my own hardware, I'm now convinced that this is > >actually normal behaviour: As far as Asterisk is concerned, a call is deemed > >"answered" as soon as the hardware seizes the line. It is only "not > >answered" > >if the line is not available. > > > >Which makes sense, because an analogue line has no D-channel. Once the trunk > >is acquired successfully, there is no way for a machine to know the state of > >the call beyond then. Such supervisory information as there is -- a regular > >cadence during ringing, possibly a burst of white noise and then a human > >voice > >-- is geared towards interpretation by human beings. > > > >Moerover, since the tones are different in every country (and sometimes, > >between different telephone exchanges in the same country; at one time, the > >UK > >was using three sets of supervisory tones depending whether you were on an > >old-fashioned "clicky-clicky" exchange, an intermediate-generation analogue > >electronic exchange or System X) it would not be a trivial task to make > >sense > >of them. > > > > > >I think if you want full supervisory information, you are going to need to > >use > >some sort of digital telephony technology (ISDN or GSM). > > > This is well known behavior for many years, since the inception of > Asterisk/Zaptel > I wonder why tests had to be run! > The OP issue was answered several days ago > His issue was obvious and well stated until another poster confused the issue!
Just to keep it clear for anyone who stumbles on this thread in the future, this can sometimes work if you set the callprogress=yes option in chan_dahdi.conf if your country/provider/exchange is supported. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
