hi all, thanks for your replies if you have 100 extensions, put them all into a single string? so: (SIP/1001&SIP/1002&SIP/1003...until you get to 100?
It is very difficult to manage such a thing, no? I don't understand the queues,ringall. can someone explain? thanks in advance On 12/05/2012 10:59 PM, Danny Nicholas wrote: > > You “can” do the queues/ringall, but you’re increasing your pay grade > by doing so. > > > > *From:*[email protected] > [mailto:[email protected]] *On Behalf Of *Carlos > Rojas > *Sent:* Wednesday, December 05, 2012 3:58 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] - configure ring group > > > > Maybe, > > > > You can do that, with queues, and ringall strategy. > > On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini <[email protected] > <mailto:[email protected]>> wrote: > > You can dial all the extensions at once, putting all them in the dial > string, separated by &. There is no other method. > > > > Leandro > > 2012/12/5 Paolo De Michele <[email protected] > <mailto:[email protected]>> > > hi all, > > I want have an information about ring group in asterisk (1.8.16 - > centos 6.3) > I have configured skypeforasterisk for incoming call to one > extension and it works > > now,my chan_skype.conf is: > > [general] > > default_user=user-skype > > [user-skype] > secret=xxxxxxxxx > context=from-skype > exten=9999 > disallow=all > allow=ulaw > allow=alaw > > my extensions.conf: > > [from-skype] > > exten => 9999,1,Verbose(2,Incoming Skype Call) > same => n,Answer() > same => n,Dial(SIP/1000&SIP/2000&SIP/3000,30) > same => n,Playback(user&is-curntly-unavail) > same => n,Hangup() > > at right time the internal ring are 1000, 2000 and 3000 > I have the extension from 1000 to 1005, 2000 to 2005 and from 3000 > to 3005 > I can ring him all? I can group the configuration into a single > string? > > let me know something > thanks in advance > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
