It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't answer your question about when it was "fixed" but I've never had a problem doing it (70 concurrent calls max, all recorded, 5 concurrent channels spied max).
On Tue, Feb 7, 2012 at 5:48 AM, Tiago Geada <[email protected]> wrote: > that means that from 1.4.18 that issue is no longer present ? > > On 7 February 2012 12:44, Jonas Kellens <[email protected]> wrote: > >> ** >> On 02/07/2012 01:07 PM, Sammy Govind wrote: >> >> Hello, >> >> I've been managing multiple call centres, almost all of them having >> their calls recorded one way or other. Even in PBX environments with >> MixMonitor and call recordings I haven't came across the situation where I >> discovered that I can't chanspy a call because its recorded ! >> Which version of asterisk you are using ! can you paste the CLI logs >> which show a complete call with a failed attempt to Chanspy ? >> >> >> Using Asterisk 1.6.2.22. >> >> The fact that ChanSpy can not be used with MixMonitor is something I read >> on the wiki : >> >> Attention >> >> - Up to and including Asterisk 1.4.17 ChanSpy can cause a * >> crash/segfault* if used together with >> Monitor<http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor>or >> MixMonitor <http://www.voip-info.org/wiki/view/MixMonitor> at the >> same time. 1.4.18 is supposed to attack this issue by using "audiohooks" >> that replaces the current ChanSpy approach. >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 > -- Carlos Alvarez TelEvolve 602-889-3003
-- _____________________________________________________________________ -- 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
