Me
2004-Dec-28 13:01 UTC
[Asterisk-Users] Sending call to analog then to Vmail after timeout?
I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the * voicemail. I can't seem to get it to timeout regardless of how many seconds I set it to. I assume this has something to do with the fact that * considers the call answered as soon as the zap channel picks it up, right? Anyhow, is there a way to make the above config work and go to the * voicemail after 20 seconds if the called party does not answer after 20 seconds? Also, what happens if the called party's line is busy, have not run into this yet so I am curious. Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041228/f0413a53/attachment.htm
Me
2004-Dec-28 13:20 UTC
[Asterisk-Users] Sending call to analog then to Vmail after timeout?
Sorry about the HTML emails, on my laptop and forgot to change the sending format from the default. ----- Original Message ----- From: Me To: asterisk-users@lists.digium.com Sent: Tuesday, December 28, 2004 2:01 PM Subject: [Asterisk-Users] Sending call to analog then to Vmail after timeout? I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number). When I do this in my extensions.conf: exten => 1200,1,playback(pls-wait-connect-call) exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) exten => 1200,3,VoiceMail(u100@lightwavetech.com) exten => 1200,4,Goto,t|1 The phone rings beyond the 20 second timeout and never really goes to the * voicemail. I can't seem to get it to timeout regardless of how many seconds I set it to. I assume this has something to do with the fact that * considers the call answered as soon as the zap channel picks it up, right? Anyhow, is there a way to make the above config work and go to the * voicemail after 20 seconds if the called party does not answer after 20 seconds? Also, what happens if the called party's line is busy, have not run into this yet so I am curious. Thanks! -- Start Your Own Internet Service! http://www.YourOwnISP.com _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
C F
2004-Dec-28 22:34 UTC
[Asterisk-Users] Sending call to analog then to Vmail aftertimeout?
---------- Forwarded message ---------- From: C F <shmaltz@gmail.com> Date: Wed, 29 Dec 2004 00:34:28 -0500 Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail aftertimeout? To: Me <mylist@lightwavetech.com> try the M option which will do a macro and will not connect the caller unless s/he presses some button. and if no button is pressed then it goes to VM. now remember to replay the message (to press the button) a few times b4 going to VM otherwise they will never hear it, since * considers it answered . http://www.voip-info.org/wiki-Asterisk+cmd+dial On Tue, 28 Dec 2004 23:29:54 -0600, Me <mylist@lightwavetech.com> wrote:> I was aware of the "c" option but it's a pain for people to have to press > the # sign plus they have to know they are suppose to do that. In addition, > I tried to use the "A" option to play a sound to them when they answer > reminding them to press pound at the end of the message but the sound > doesn't play until they press pound :) > > So.. It appears I am still stuck with * considering the call answered when > the Zap channels grabs it and connects the other leg of the call. Hopefully > there is some other way to make this happen. > > Thanks for the feedback though. > > Start Your Own Internet Service! > http://www.YourOwnISP.com > > ----- Original Message ----- > From: "C F" <shmaltz@gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Tuesday, December 28, 2004 6:26 PM > Subject: Re: [Asterisk-Users] Sending call to analog then to Vmail > aftertimeout? > > > Follow these: > > http://www.voip-info.org/wiki-Asterisk+zap+channels > > looks like this would work: > > exten => 1200,1,playback(pls-wait-connect-call) > > exten => 1200,2,Dial(Zap/1c/5555551212,20,rTt) ;note the c after the > > channel number > > exten => 1200,3,VoiceMail(u100@lightwavetech.com) > > exten => 1200,4,Goto,t|1 > > > > > > On Tue, 28 Dec 2004 14:20:02 -0600, Me <mylist@lightwavetech.com> wrote: > > > Sorry about the HTML emails, on my laptop and forgot to change the > sending > > > format from the default. > > > > > > > > > ----- Original Message ----- > > > From: Me > > > To: asterisk-users@lists.digium.com > > > Sent: Tuesday, December 28, 2004 2:01 PM > > > Subject: [Asterisk-Users] Sending call to analog then to Vmail after > > > timeout? > > > > > > I have one analog line hooked in my Asterisk box using an x100p (I think > > > that's the model number). > > > > > > When I do this in my extensions.conf: > > > > > > exten => 1200,1,playback(pls-wait-connect-call) > > > exten => 1200,2,Dial(Zap/1/5555551212,20,rTt) > > > exten => 1200,3,VoiceMail(u100@lightwavetech.com) > > > exten => 1200,4,Goto,t|1 > > > > > > The phone rings beyond the 20 second timeout and never really goes to > the * > > > voicemail. I can't seem to get it to timeout regardless of how many > seconds > > > I set it to. > > > > > > I assume this has something to do with the fact that * considers the > call > > > answered as soon as the zap channel picks it up, right? > > > > > > Anyhow, is there a way to make the above config work and go to the * > > > voicemail after 20 seconds if the called party does not answer after 20 > > > seconds? Also, what happens if the called party's line is busy, have not > run > > > into this yet so I am curious. > > > > > > Thanks! > > > > > > -- > > > Start Your Own Internet Service! > > > http://www.YourOwnISP.com > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > 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 > > > Asterisk-Users@lists.digium.com > > > 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 > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Warren Burstein
2004-Dec-29 15:09 UTC
[Asterisk-Users] Sending call to analog then to Vmail after timeout?
I came across the same problem today. Phase one of our * project is to replace the PBX in one of our offices with *, and one of the extensions will be sent over VOIP to service representatives at a different location. But as a fallback, we want to dial directly if VOIP doesn't work (maybe the network is down) and if that doesn't work, send the caller to voicemail. And the same problems are happening. I'm not sure if shmaltz's solution helps us here, during working hours at the place where the service reps sit, it will go into a PBX (not *), I think it will ring on the rep's desk, using Pavlovian techniques (e.g. send the ring voltage to their chair if they don't press a key), they can be trained to hit a key when they pick up the phone. But when the help desk isn't staffed, we want to ring someone's cellphone. Which is fine if he answers, but if he misses the call, or is already taking a call, we want it to go to his cellphone's voicemail (which is more accessable to him outside the office than his voicemail on the * server). Maybe we can put a DTMF digit into the cellphone voicemail greeting. On the other hand, maybe hitting any key while recording the greeting will stop recording. Then I was thinking about phase two of our project. In our main office, we currently have a PBX which we're not planning to replace, but we do want to replace our IVR/voicemail system that runs on Windows using a Dialogic card with an * system, the dialogic picks up incoming calls, transfers them (sending the PBX a flash followed by a local extension) to people, and detects if they answer or not. The Dialogic card knows how to detect ring, busy, no answer, no dialtone, congestion tone, and so on, so if the callee does not answer the call, it goes to voicemail (or to a different person, or back to the main menu so the caller can try someone else). It looks to me like we would have to have everyone in the office press a key every time any phone rings, I'm not sure if that's acceptable. Would call progress be any help? From what I've read, it can sometimes cause calls to be disconnected.