Re: [asterisk-users] Simple Meetme Question
2009/3/8 Sven Geggus use...@fuchsschwanzdomain.de Gavin Henry gavin.he...@gmail.com wrote: Just transfer them to your meetme extension after you've called them. Hm, how would I do this? Until now call switching usually ended for me when the call has been established. I'm using a SIP phone connected to an asterisk box which is connected to the world via various ways (ISDN, SIP, IAX2). So what would I do on the my SIP phone after the call has been established and what needs to be changed in the dialplan to actually reconnect the current call to the MeetMe Conference then? Sven You need to transfer option enabled in dial() (tT) CLI core show application Dial And you need to press a transfer button ;) ___ -- 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
[asterisk-users] Simple Meetme Question
Hello, setting up Meetme was very easy. I jut added the MeetMe Application to an internal extension to be reachable by SIP and to an external extension to be reachable by ISDN. What I don't understand however is how to call somebody and drop him to the conference? I'm using Asterisk 1.4 from Debian lenny Sven -- In the land of the brave and the free, we defend our freedom with the GNU GPL (Richard M. Stallman on www.gnu.org) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- 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
Re: [asterisk-users] Simple Meetme Question
Just transfer them to your meetme extension after you've called them. Just like you would transfer someone who has called you. * will then put them into that conference. Thanks. On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote: Hello, setting up Meetme was very easy. I jut added the MeetMe Application to an internal extension to be reachable by SIP and to an external extension to be reachable by ISDN. What I don't understand however is how to call somebody and drop him to the conference? I'm using Asterisk 1.4 from Debian lenny Sven -- In the land of the brave and the free, we defend our freedom with the GNU GPL (Richard M. Stallman on www.gnu.org) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- 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
Re: [asterisk-users] Simple Meetme Question
Gavin Henry gavin.he...@gmail.com wrote: Just transfer them to your meetme extension after you've called them. Hm, how would I do this? Until now call switching usually ended for me when the call has been established. I'm using a SIP phone connected to an asterisk box which is connected to the world via various ways (ISDN, SIP, IAX2). So what would I do on the my SIP phone after the call has been established and what needs to be changed in the dialplan to actually reconnect the current call to the MeetMe Conference then? Sven -- The main thing to note is that when you choose open source you don't get a Windows operating system. (from http://www.dell.com/ubuntu) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- 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
