Hi, thanks for your time!
O Venres, 5 de Agosto de 2011 12:35:05 escribiches: > Completely normal operation. > You need to read and understand more basic telephony and analog lines to > understand why that won't work. I definitely have a lot to learn yet. > Asterisk needs to be in control, and once someone answers a phone not under > Asterisk control, or the call is abandoned there is little you can do. What I pretend is that asterisk detects that it's not under control and gets out of the way. The same way it detects a remote hangup and stops the dialplan, it could detect that someone else answered (the line is not ringing anymore) and discard it the same way it does when the remote part hangup. I've read comments in forums and tutorials that seem to imply that this happens, but I couldn't find any confirmation (and indeed, it's not happening to me). If you confirm me that this is the normal behavior, then I at least I know my solution is in the dialplan and not a card/line/driver problem. > Sounds like a task for a simple answering machine from Wal-Mart > All you other phones should be connected to FXS ports, or you need to be > smarter in your dialplan. Once you answer, Asterisk is behaving normally Yes, it's a really simple task, but this should be just a starting point. The plan is to start migrating services to the PBX little by little, and the voicemail looked like the easier thing to start. I wanted to maintain the current analog phones until I feel confident with the asterisk configuration. Maybe it wasn't such a great idea, and I should start by moving the phones to FXS ports in the PBX. > > John Novack > > Jorge Barreiro wrote: > > Hi again, > > > > thanks for your answer, but it didn't solve the problem. That Dial > > command returns inmediately, so I don't even have the delay. > > > > I'll try to explain myself better. The PBX has only one FXO card, > > connected to the PSTN. There is no other phones connected to the PBX nor > > SIP extensions. There are analog phones connected to the same PSTN. > > > > What I try to do is that, when there is an incoming call from the ouside, > > if someone answers on a phone, then the PBX won't answer. > > > > > > Thanks. > > > > O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu: > >> Hi, > >> > >> your concept using Wait() won't work here. > >> Try it like this: > >> > >> [incoming] > >> exten => s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s > >> exten => s,n,BackGround(wellcome-message) > >> exten => s,n,Voicemail(1234) > >> exten => #,1,Hangup() > >> > >> So, of you answer the call within 30s, you'll get the call on your > >> phone. After 30s, the Voicemail will answer the phone. > >> > >> > >> regards, > >> Ruben > >> > >> Am 04.08.2011 21:39, schrieb Jorge Barreiro: > >>> Hello, > >>> > >>> I'm configuring an Asterisk PBX to use as an answering machine. I have > >>> a FXO card connected to the line, and other analog telephones > >>> connected to the same line. The PBX answers and redirects you to the > >>> voicemail after a delay. > >>> > >>> The problem is that even if I pickup any analog phone in the line > >>> before the PBX does, it answers after the delay anyway. And I couldn't > >>> find how to prevent this, or even if this is supposed to happen. > >>> > >>> My FXO card is a cheap X100P (source of problems, I know), and I'm > >>> using the Asterisk version included in Debian Squeeze (1.6.2.9). > >>> My dial plan looks like this: > >>> > >>> [incoming] > >>> exten => s,1,Wait(8) > >>> exten => s,2,Answer > >>> exten => s,3,BackGround(wellcome-message) > >>> exten => s,4,Voicemail(1234) > >>> exten => #,1,Hangup > >>> > >>> I don't know if this is related, but I'm in Spain and I had to add: > >>> hanguponpolarityswitch=yes > >>> to the chan_dahdi.conf file so that asterisk detects the remote hangup. > >>> I also added: > >>> answeronpolarityswitch=yes > >>> but this didn't help. It seems to be used just to detect the answer > >>> when you are calling, not when receiving a call. > >>> > >>> > >>> I'd appreciate any help you could provide. > >>> > >>> Thanks! > >>> > >>> -- > >>> _____________________________________________________________________ > >>> -- 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 > >> > >> -- > >> _____________________________________________________________________ > >> -- 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 > > > > -- > > _____________________________________________________________________ > > -- 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 -- _____________________________________________________________________ -- 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
