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.Love the idea. How? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281
> > On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk.org at sedwards.com> wrote: > > 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. > > Love the idea. How?exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3) -- 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/20170206/978a4065/attachment.html>
It was a very long time ago, so I'd have to dig through some old notebooks to get the exact details, but it wasn't too difficult. Basically, two calls are made. The first call is made with the wait time set to about a quarter of a second. We modified the Asterisk source code to allow floating point values for the wait time, basically modifying only 1 or 2 lines of code. Even a non-programmer could do it. When the first call is made for such a short period, the remote end still goes off hook, but the call will end before it starts to ring. Then, halfway through the first call, a second call is made. Since the remote end is off hook from the first call, the second call will get sent to voicemail and the message is played there. I remember having to do some testing to get optimal wait times and delays in milliseconds, but overall, the ability to go straight to voicemail was a valuable tool for us. Regards; John V. -----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 11:35 AM 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.Love the idea. How? -- 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
> 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
On 6/2/17 5:24 pm, Tech Support wrote:> Basically, > two calls are made....>When the first call is made for > such a short period, the remote end still goes off hook, but the call will > end before it starts to ring. Then, halfway through the first call, a second > call is made. Since the remote end is off hook from the first call, the > second call will get sent to voicemail and the message is played there.Am I right in thinking call waiting isn't a thing on US mobile networks then? In the UK, call waiting is pretty standard, and almost universally enabled by default on mobile networks. AIUI the same is true for much of Europe. Kind regards, Chris -- This email is made from 100% recycled electrons