Re: [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer
So: A calls B B answers B puts A on hold B calls C B talks to C B ends conversation with C B talks to A again, regardless I this correct? Looks like a simple "Hold" exercise. jg -- _ -- 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
[asterisk-users] terminating the call, when transferer hangs up the call during attended transfer
Hello guys, I'm vainly trying to figure out how to setup quite a strange customers requirement: they require that when during attended transfer, (A->B->C) whenever B hangs up the call before it's connected to C, the call just returns to B, instead of changing to blind transfer. I tried using atxfer drop call option (enabling null channel in sources), but to no avail.. Is there some way how this could be achieved? thanks a lot for any help.. with best regards nikola ciprich -- - Ing. Nikola CIPRICH LinuxBox.cz, s.r.o. 28.rijna 168, 709 00 Ostrava tel.: +420 591 166 214 fax:+420 596 621 273 mobil: +420 777 093 799 www.linuxbox.cz mobil servis: +420 737 238 656 email servis: ser...@linuxbox.cz - pgphgODqGyj4G.pgp Description: PGP signature -- _ -- 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
[asterisk-users] combine external video source and audio call to make SIP video call?
I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of graphical SIP client; ideally this should all be done headlessly in the Asterisk dialplan. Any suggestions on where to start? -- Eric Cooper e c c @ c m u . e d u -- _ -- 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
Re: [asterisk-users] DAHDI Missing '/sys/bus/astribanks/drivers/xppdrv/sync'
On Sat, Nov 23, 2013 at 11:47:49AM -0800, Joseph Towery wrote: > Hello group, > > I am installing Asterisk on a new pc running Ubuntu 12.04.3 > Server. I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried > 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. I am following along in the > ASTERISK The Definitive Guide 4th Edition and on page 116 when I > issue the command $ sudo /etc/init.d/dahdi start I get the > following error: > > Loading DAHDI hardware modules: > wctdm24xxp: done > Running dahdi_cfg: done. > Missing '/sys/bus/astribanks/drivers/xppdrv/sync' Hi, it's safe to ignore this message, but for future reference I just committed a change [1] that will now prevent it from showing, which will be in the final v2.8.0 release of DAHDI-tools. [1] http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=commit;h=c27664dc1b3 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
