hi Leandro, thank to you for your reply but I think I shall apply the configurations that advised me AJS many thanks for your help cheers
On 12/06/2012 09:13 AM, Leandro Dardini wrote: > 100 extension on a row is not feasible... the queue strategy is the > only possible solution. If you check the queue.conf file you'll find > you can define a "Queue" and add as many members you like. One of the > strategy available is the "Ring all" where all the members in the > queue will be ring. You can let your peers to log in/log out of the > queue via dialplan > > Leandro > > 2012/12/6 Paolo De Michele <[email protected] > <mailto:[email protected]>> > > 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]> >> [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 > > > > > -- > _____________________________________________________________________ > -- 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
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