[asterisk-users] Talk detection during call
I am looking for ways to detect if there is some person talking on the other side of the line and trigger some events based on that.. is there any possible way through which this could be done in asterisk ? Thanks, Sathiish -- _ -- 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] passing arguments to macros from originate command
I wanna be able to pass arguments to macros when i initiate a call through the originate command.. Is there any possible way of passing arguments to the originate command in some way ? -- _ -- 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] wrong RTT QoS information always reported
I am using CHANNEL function to store the rtt as variable in cdr.When i checked the records i found that rtt was always zero.To double check i turned on rtcp debug I was only able to see sent rtcp,sending rtcp but no got rtcp. Also the timestamps in rtp debug were oddly dissimilar.Is there a prblem wth my timestamp or is this a known bug? -- Sathiish Kumar Mohan Kumar Technical Operations Intern, Citrix Online 7408 Hollister Avenue,Goleta, Santa Barbara CA-93106 - -- _ -- 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] Maximum concurrent calls using call files
I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length) Thanks, Sathiish -- _ -- 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] Maximum concurrent calls using call files
I've previously used iperf for my project and It can only simulate TCP/UDP traffic.and the thing is I'm testing this on a platform which does only RTP/SIP.I am not sure they've this facility.Anyhoo i wanted to know if it was possible to make such concurrent calls using Asterisk On Fri, Jul 6, 2012 at 3:00 PM, Stephen J Alexander sjalexan...@mpbx.comwrote: I haven't used it, so can't recommend it per se; but as I understand it, iperf is a tool that can do that kind of simulation for you: http://iperf.sourceforge.net/ might be worth trying before you build your own modules. Regards, Stephen J Alexander MPBX, LLC http://mpbx.com 832-713-6729 On Fri, Jul 6, 2012 at 4:01 PM, sathiish kumar sathiish.ku...@gmail.com wrote: I am planning on building a testing module which would spawn about 500 calls in order to test the performance of the network by transferring audio/speech files to end points at that juncture.Is it possible to spawn as many concurrent calls (or nearly concurrent calls) using just call files.Is there a limit as to the maximum number that could be spawned.? I tried doing this for about 20 calls and found that there is autofallthrough after a point of time.Is this a problem with my dialplan or is it because of the call files (i also get a warning which states that the ast_queue_frame:Exceptionally long queue length) Thanks, Sathiish -- _ -- 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 -- _ -- 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 -- _ -- 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] Outbound Asterisk calls default directmedia specifications
I am using call files to make calls to a remote machine but can't seem to quite understand the directmedia options that are set by default in Asterisk.Is there any way i can specify the directmedia options using call files? -- _ -- 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
