you add a line like this to your dialplan <action application="bridge" data="portaudio/auto_answer"/> On Fri, Sep 11, 2009 at 8:49 AM, vip killa <vipkilla@gmail.com> wrote:
what i'm trying to do is have an incoming call be answered by port audio without having to do a pa answer? i don't see how this is possible using originate.
On Thu, Sep 10, 2009 at 7:36 PM, Michael Collins <msc@freeswitch.org>wrote:
On Thu, Sep 10, 2009 at 4:34 PM, Michael Jerris <mike@jerris.com> wrote:
portaudio/auto_answer dial string
On a similar theme, check this one out: http://wiki.freeswitch.org/wiki/Dialplan_XML#Getting_Fancy_With_PortAudio -MC
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
_______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400