Thanks for the response. Have you try the new TDM FXO cards? Does call progress work with those?
----- Original Message ----- From: "Vic Cross" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, May 13, 2004 5:46 AM Subject: Re: [Asterisk-Users] problems with analog interface to PBX > On Wed, 12 May 2004, Dan Fernandez wrote: > > > Asterisk should answer the call, playback a message, dial another PBX > > extension and if no one answers dial another extension (via IAX). > > > > The first problem I ran into was that the Flash application doesn't > > really work. To get around this I added another x100p to dial the new > > extension. The problem I ran here was that even though I specified in > > the Dial app to just dial for 30 seconds, it rang forever as if * cannot > > recongnize that no one had picked up. Asterisk does seem to detect > > hangups and busy tones (I have busydetect=yes and busycount=10) > > In the absence of call progress detection settings, Zap analog channels > tell Dial() that they are Connected more-or-less as soon as they have > completed dialling (I see this on the display of my 7960: I see Proceeding > for a second or two, then Connected, when I dial through an X100P). So, > the timeout on your Dial() never gets triggered because the channel > reports a connected call almost straight away. > > To do what you want, you would need callprogress=yes -- as long as your > Panasonic PBX generates authentic US tones. busydetect will only detect > busy (!), not ringback or congestion or any of the other tones you would > need to make your application work the way you want -- call progress > detection tries to do this for you. > > The bad news is that even if your PBX generates US tones, reports are that > the detection is not too reliable. > > > Am I trying to do something that the x100p is not capable of? Would > > making changes to the indications.conf help at all? > > It's not that the X100P can't do the job, it's more that analogue lines > can't do the job :) Seriously, if your PBX generates US tones then give > callprogress=yes a try. From my reading of the code, the tones specified > in indications.conf are unrelated to the way the * DSP does call progress > detection (have a look at functions like ast_dsp_call_progress() in dsp.c > if you're really curious). > > > 2) I would also like to use * for voicemail. The user should be able to > > dial the extension where the x100p is connected and asterisk recognized > > the extension the user is dialing and request for the password? Is this > > possible? > > On an analogue channel via an X100P, there is no "called number" > indication. So you can't tell what number the caller dialled to reach > you. If you wanted to use the * box as a voicemail-only machine, you > could drop the caller straight into VoiceMailMain, but if you wanted other > functions (conference rooms, VoIP gateway, etc) you would need to use an > IVR... > > "press 1 to access Voicemail... > press 2 to reach a Voice-over-IP user... > press 3 to join a conference... > ..." > > This doesn't really help your original need: to dial another number on the > PBX and get control back if needed. If callprogress=yes doesn't work for > you, you could try something like this (off the top of my head): > > exten => 4,1,Playback(trying-press-*-to-come-back) > exten => 4,2,Dial(Zap/1/1234,,Hg) > exten => 4,3,Goto(103) > exten => 4,103,Playback(sorry-cant-reach) > exten => 4,104,Goto(menu,s,1) > > On the Dial(), the option H enables caller hangup using '*', and g says go > on in context when the destination channel hangs up. This would put your > caller in the driver seat and get them to do the tone detection for you ;) > > > Hope this helps, > Vic Cross > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users