> -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neo haux > Sent: Monday, 6 May 2013 1:34 p.m. > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Joining an astablished call > > Hi, > > I don't know how to call this functionality, but what I want > to do is join an already established communication between > PSTN---FXS_connected_phone using my SIP phone (I have an > asterisk v11 with digium TDM400P at home) > > Is it possible? What I don't want is using the conference > sound and menu.... It's just a normal call between to > channels that I have to join for few minutes. > > Regards > > >
exten => 1234,1,ChanSpy(SIP/cisco1,qn) Assuming cisco1 is a sip extension. I haven't tried it but below should work. exten => 1234,1,ChanSpy(DAHDI/1,qn) Alec -- _____________________________________________________________________ -- 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
