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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130505/30d983fb/attachment.htm>
On 05/05/2013 08:34 PM, neo haux wrote:> 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)I had this set up once upon a time. It took quite a bit of dialplan hackery, but the basic idea is to create a conference bridge, use ChannelRedirect() to connect both existing channels to the bridge, and then join it yourself. This worked for a bit, but the last time I tried it, Asterisk segfaulted. Oops. -- =======================================================================Ian Pilcher arequipeno at gmail.com Sometimes there's nothing left to do but crash and burn...or die trying. ========================================================================
> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of neo haux > Sent: Monday, 6 May 2013 1:34 p.m. > To: asterisk-users at 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
In the telephony world that is known as "barge-in" and is a programmable option granting that right to specific extension(s) in systems that normally have automatic privacy. Not all electronic key and hybrid systems have automatic privacy, though most do. John Novack neo haux wrote:> 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 > > > > > > -- > _____________________________________________________________________ > -- 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-- Dog is my Co-pilot -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130506/4a04cd8a/attachment.htm>