Thank you for your interesting comments.
On Sat, Feb 6, 2010 at 4:14 PM, Erik de Wild: Tripple-o <i...@tripple-o.nl> wrote: > Thomas, > > Yes you can do this. I actually have done this and run it as a > service under the name Meetmecall. I use MSN as the user interface to > record the message, create phone lists of the numbers that has to be > called and to actually schedule and perform the delivery. It is > possible to use it for spam but the customers I have use it to notify, > remember, offer or let the callee know about something relevant, but > always as part of an already existing relation. With some extra > parameters used, you can start a groupcall and use all the other > Asterisk magic available like doing a questionarry using a smart IVR > etc. etc. I can think about a long list of useful use of this service. > > I have no idea about the rules and legislation in other countries but > in Holland you will end up with serious trouble and extreme high > penalties to pay if you actually use it for spamming. > > I will not send you a copy of the solution but it is based on the use > of call files pointing to local channels/extensions where the Asterisk > magic is combined in a working (and I think clever) way. The CDR isn't > perfect but disable and enable CDR at the proper points in the dial > plan and clever use of the USERFIELD variable will result in useable > data for billing the users. The CDR shows that most callees, listen to > the message until it ends and yes, sometime there are complaints about > the use but that is very rare. > > About the scheduling of the calls to make. It is not Asterisk that > limits you. Far before reaching the limits of Asterisk it will be the > bandwidth available and the SIP trunk provider that normally doesn't > allow an endless number of concurrent calls. When I started this I was > working for a Norwegian company offering the dial tone on the internet > and I had a server almost directly connected to the backbone of > internet with more or less endless bandwidth. I did some stress > testing of a call center solution and 80 concurrent calls wasn't a > problem and my guess is that you can far beyond 80 calls. It is wise > to move the call files one after the other or one batch after the > other. Moving large numbers of call files into /var/spool/asterisk/ > outgoing might sometimes result in unexpected and not intended > results. There are other scenarios but this was my choice. > > 10.000 calls will take some time but with a 30 seconds message, 20 > concurrent calls and 10 seconds average to dial after around 5,5 hours > the last phone call will be dialed. If the message is just 15 seconds > it will take around 3,5 hours. If you want to deliver in short time, > like 10 minutes, you really have to scale up to 420 concurrent calls > which doesn't sound doable unless you have real serious budgets. If > you put everything in place at your side you will probably run into > constraints of the SIP provider and the interconnection to the pstn. > > btw: > Asterisk has the potential to build lots of evil features and lots of > standard features can be used in an evil way. Personally I think it is > kind of strange that, if a question is asked, one has to explain why > the answer is not mend for evil use. We don't have to help someone out > and we can refuse because of lots of reasons: no time, not an > interesting question, not a single sign of any effort by the one > asking the question, not willing to give something away that costs > lots of time and energy, the feeling that it will be used in an evil > way etc. etc. I think the tone and the content of this discussion > harms the Asterisk community as a whole. > > with friendly regards, > > > Erik de Wild > Tripple-o: your asterisk migration partner > the Netherlands > > > > > > > > On 6 feb 2010, at 03:54, Thomas Perron wrote: > >> Does anyone have a Dial script or a hint on how I can dial 10000 >> numbers in sequence? >> When the calls are answered, I play a .gsm or .wav. >> Then, if user presses a defined digit, the call gets bridged to me. >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users