Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt> On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote: > > That's the basics, but you have to nail the timing just right. The timing is > really important to do it the right way. > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards > Sent: Monday, February 06, 2017 12:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call List Campaign to an IVR > > >> On Mon, 6 Feb 2017, Tech Support wrote: >> >> We were able to develop a feature to send the call to voicemail > about 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) >> delete the message without listening to it, or (3) listen to the > message when it was most convenient for them. That way, they were in control > and things were done on >> their terms. > >> On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> Love the idea. How? > >> On Mon, 6 Feb 2017, Matt Riddell wrote: >> >> exten => >> _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3) > > Amazing. Who knew? > > So how/why does this work? > > I see 2 calls going out to my cell. Does the first 'busy out' my number at > my cell provider so the second goes straight to VM? What part does the > '0111' play? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170206/d965b50f/attachment.html>
I remember doing the testing and two calls going out at the same time don?t actually have to go out at the *exact* same time. The remote end will pick up one of the two calls, but there is no guarantee which one it will be. Also, if you let the first call ring too long, yes, the second call will go to voicemail, but the first call will start ringing, which is something we wanted to avoid. John From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt Riddell (lists) Sent: Monday, February 06, 2017 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call List Campaign to an IVR Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote: That's the basics, but you have to nail the timing just right. The timing is really important to do it the right way. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, February 06, 2017 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call List Campaign to an IVR On Mon, 6 Feb 2017, Tech Support wrote: We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2) delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were done on their terms. On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk.org at sedwards.com> wrote: Love the idea. How? On Mon, 6 Feb 2017, Matt Riddell wrote: exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3) Amazing. Who knew? So how/why does this work? I see 2 calls going out to my cell. Does the first 'busy out' my number at my cell provider so the second goes straight to VM? What part does the '0111' play? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170206/5c81e0ef/attachment.html>
there are providers which let you call directly to voicemail by using a prefix On Mon, Feb 6, 2017 at 8:28 PM, Tech Support <asterisk at voipbusiness.us> wrote:> I remember doing the testing and two calls going out at the same time > don?t actually have to go out at the *exact* same time. The remote end will > pick up one of the two calls, but there is no guarantee which one it will > be. Also, if you let the first call ring too long, yes, the second call > will go to voicemail, but the first call will start ringing, which is > something we wanted to avoid. > > John > > > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] *On Behalf Of *Matt Riddell (lists) > *Sent:* Monday, February 06, 2017 12:32 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Call List Campaign to an IVR > > > > Not really, doing the way below you don't even have to worry about it. > They both go out at the same instant and as soon as it hits voicemail it > disconnects the other leg. > > > > If you wanted you could leave it ringing for twenty minutes and it would > still have the same effect. > > Kind regards, > > > > Matt > > > On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> > wrote: > > That's the basics, but you have to nail the timing just right. The timing > is > really important to do it the right way. > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com > <asterisk-users-bounces at lists.digium.com>] On Behalf Of Steve Edwards > Sent: Monday, February 06, 2017 12:25 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call List Campaign to an IVR > > > > On Mon, 6 Feb 2017, Tech Support wrote: > > > > We were able to develop a feature to send the call to voicemail > > about 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) > > delete the message without listening to it, or (3) listen to the > > message when it was most convenient for them. That way, they were in > control > and things were done on > > their terms. > > > > On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk.org at sedwards.com> > > wrote: > > > > Love the idea. How? > > > On Mon, 6 Feb 2017, Matt Riddell wrote: > > > exten => > > _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3) > > > Amazing. Who knew? > > So how/why does this work? > > I see 2 calls going out to my cell. Does the first 'busy out' my number at > my cell provider so the second goes straight to VM? What part does the > '0111' play? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 > <(760)%20468-3867> PST > https://www.linkedin.com/in/steve-edwards-4244281 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170206/39b00a44/attachment.html>
> On 6/02/2017, at 1:28 PM, Tech Support <asterisk at voipbusiness.us> wrote: > > I remember doing the testing and two calls going out at the same time don?t actually have to go out at the *exact* same time. The remote end will pick up one of the two calls, but there is no guarantee which one it will be. Also, if you let the first call ring too long, yes, the second call will go to voicemail, but the first call will start ringing, which is something we wanted to avoid. > John >That's the benefit of doing the & thing. The instant one of them goes to voicemail the other will stop ringing. Typing calls this happens in a few ms (after post dial delay). Because they are both going out at the same time with the same provider this is super quick. -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170206/2c543101/attachment.html>