Roi Stork
2008-Oct-27 05:19 UTC
[asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)
Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the agents notice that the other line is silent. The queue is already set up to hold an infinite number of calls (meaning: maxlen=0/no limit), and the agents are already answering the calls immediately/after one ring, but the problem still shows up. Is forwarding to a meetme conference room faster than through a queue? On Thu, Oct 16, 2008 at 11:25 PM, Steve Totaro < stotaro at totarotechnologies.com> wrote:> If you can figure out how to generate .call files from your DB > entries, you have it made. > > Vicidial needs alot of work as far as I am concerned, for free it is > OK I guess. I think using meetme conference rooms for everything is a > kludgy hack, and the UI is less than nice (if you are into UIs). > > I suggest you continue on your own custom development if you have the > time. Check out Aheeva for inspiration. > > Thanks, > Steve Totaro > > On Fri, Oct 17, 2008 at 1:31 AM, ram <talk2ram at gmail.com> wrote: > > look at Vicidial > > > > ram > > > > On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <yvzyldrm at gmail.com> > wrote: > >> > >> hi everybody > >> > >> This is Yavuz YILDIRIM > >> > >> I am software developer.I have a some problems in asterisk. > >> I am using mysql db. Realtime using asterisk modules. On db i am using > >> calling hundred fields for use dial. > >> But i don't know how i can automaticly dial this fields on records > >> numbers. Who can help me asterisk api and others. > >> > >> Thank you > >> > >> > >> _______________________________________________ > >> -- 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 > > > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081026/a6c93f35/attachment.htm
Roi Stork
2008-Oct-31 03:41 UTC
[asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)
Additional question: are there instances when the incoming call waiting in the queue is dropped when connected to a waiting agent/local extension? By the way, incoming call channel is: Local/XXXXXXXXX at context created via Originate On Sun, Oct 26, 2008 at 10:19 PM, Roi Stork <roi.stork at gmail.com> wrote:> Also posting this question to people working on manager interface and > dialers. > > I have a simple auto dialing script (using Originate) that forwards all > incoming calls to a queue full of waiting agents instead of a meetme > conference room. I use queues rather than meetme so I can leave the > automatic call distribution to the queue itself. > > The problem is when the calls reach the agents, some of the agents notice > that the other line is silent. The queue is already set up to hold an > infinite number of calls (meaning: maxlen=0/no limit), and the agents are > already answering the calls immediately/after one ring, but the problem > still shows up. > > Is forwarding to a meetme conference room faster than through a queue? > > On Thu, Oct 16, 2008 at 11:25 PM, Steve Totaro < > stotaro at totarotechnologies.com> wrote: > >> If you can figure out how to generate .call files from your DB >> entries, you have it made. >> >> Vicidial needs alot of work as far as I am concerned, for free it is >> OK I guess. I think using meetme conference rooms for everything is a >> kludgy hack, and the UI is less than nice (if you are into UIs). >> >> I suggest you continue on your own custom development if you have the >> time. Check out Aheeva for inspiration. >> >> Thanks, >> Steve Totaro >> >> On Fri, Oct 17, 2008 at 1:31 AM, ram <talk2ram at gmail.com> wrote: >> > look at Vicidial >> > >> > ram >> > >> > On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim <yvzyldrm at gmail.com> >> wrote: >> >> >> >> hi everybody >> >> >> >> This is Yavuz YILDIRIM >> >> >> >> I am software developer.I have a some problems in asterisk. >> >> I am using mysql db. Realtime using asterisk modules. On db i am using >> >> calling hundred fields for use dial. >> >> But i don't know how i can automaticly dial this fields on records >> >> numbers. Who can help me asterisk api and others. >> >> >> >> Thank you >> >> >> >> >> >> _______________________________________________ >> >> -- 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 >> > >> >> >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> +12024369784 (Skype) >> >> _______________________________________________ >> -- 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081030/4308fa63/attachment.htm