Simply, You can use Originate command like
originate SIP/151 application Meetme 1234,dcs if you want to dial multiple extension then just use while loop . regards Dhaval On Wed, Oct 28, 2009 at 6:45 PM, Danny Nicholas <da...@debsinc.com> wrote: > Mea Culpa?? Since I’ve only been dabbling with AMI for about 6 weeks, I > hadn’t stumbled upon the Async parameter. A “more correct” dissertation of > the sentence would be > > “The AMI originate by default operates in a synchronous or threaded > fashion, unless you specify Asynchronous mode using Async: true”. Guess > I’ll never be as smart as you, Matt. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria > *Sent:* Wednesday, October 28, 2009 5:58 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] How to dial multiple extensions at once > likeinaring group and put them in conference? > > > > Hi Matt, > > That is exactly what I am doing now and it has solved my problem. Now all > the calls originate instantly with no noticeable delay. > > -- > Zeeshan A Zakaria > > On Wed, Oct 28, 2009 at 12:18 AM, Matt Riddell <li...@venturevoip.com> > wrote: > > On 28/10/09 3:52 AM, Danny Nicholas wrote: > > This might be a better application of a call file than an AMI originate. > > The AMI originate in this case has to operate in a threaded fashion, > > whereas if you created a call file for each extension and dumped them > > into /var/spool/asterisk/outgoing, pbx.c would call all of them at once > > without the “first pickup” problem. > > Not true - you can use Async mode in an Asterisk Manager originate > command to create a call and return instantly. > > -- > Cheers, > > Matt Riddell > Director > _______________________________________________ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) > http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) > > > _______________________________________________ > -- 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 >
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