Hi
Asterisk support dialout conference?.My requirement is when type a
CLI command with argument as a number ,asterisk should able to make a
call to that number and when connected ,that channel should entered in
to the conference room,like this I should able to add multiple users
into the conference.I am using ConfBridge application for asterisk
version 1.6.2
Thanks
Nikhil
On 06/15/2011 01:34 AM, Nikhil wrote:> Hi > Asterisk support dialout conference?.My requirement is when type a CLI > command with argument as a number ,asterisk should able to make a call > to that number and when connected ,that channel should entered in to > the conference room,like this I should able to add multiple users into > the conference.I am using ConfBridge application for asterisk version > 1.6.2This is something that can be accomplished with the manager interface or call files. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
Hi, You may used the Page() function of asterisk. Which will work the same as you are required at this moment. On Wed, Jun 15, 2011 at 12:51 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> On 06/15/2011 01:34 AM, Nikhil wrote: > >> Hi >> Asterisk support dialout conference?.My requirement is when type a CLI >> command with argument as a number ,asterisk should able to make a call >> to that number and when connected ,that channel should entered in to >> the conference room,like this I should able to add multiple users into >> the conference.I am using ConfBridge application for asterisk version >> 1.6.2 >> > > This is something that can be accomplished with the manager interface or > call files. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _____________________________________________________________________ > -- 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 >-- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110615/4248ade5/attachment.htm>
Thanks for the helps I use channel originate command to achieve this. Command: asteriskCLI> channel originate SIP/201 application ConfBrigde 1234 This will make a call to the 201 user and when connected,it will be routed to conference room . Thanks NIkhil On 06/15/2011 02:17 PM, virendra bhati wrote:> Hi, > > You may used the Page() function of asterisk. Which will work the same > as you are required at this moment. > > > > On Wed, Jun 15, 2011 at 12:51 PM, Alex Balashov > <abalashov at evaristesys.com <mailto:abalashov at evaristesys.com>> wrote: > > On 06/15/2011 01:34 AM, Nikhil wrote: > > Hi > Asterisk support dialout conference?.My requirement is when > type a CLI > command with argument as a number ,asterisk should able to > make a call > to that number and when connected ,that channel should entered > in to > the conference room,like this I should able to add multiple > users into > the conference.I am using ConfBridge application for asterisk > version > 1.6.2 > > > This is something that can be accomplished with the manager > interface or call files. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > -- > _____________________________________________________________________ > -- 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 > > > > > -- > > > > ----- > Thanks and regards > > Virendra Bhati > +91-9172341457 > Asterisk Engineer > > > -- > _____________________________________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110615/54502160/attachment.htm>