Mixmonitor also muxes the two sides of the conversation after hangup. That is quite a bit of I/O for 60 simultaneous calls lasting an average of 5-15mins
On Wed, Jan 2, 2013 at 9:59 AM, Steve Totaro <[email protected]> wrote: > It depends on what you do with them. > > Years ago, 60 calls would start to crap out audio on live calls and I > learned that the hard way on a production call center. There was the > I/O of just SLIN, then converting to MP3, then transferring to a not > too forgiving SAMBA share. Scheduling things for a slower times and > moving the MP3 conversion to the mass storage significantly helped > while scrambling to find the permanent solution. > > People could increase those numbers with RAMDisk and other tricks but > just moving it off the "Phone System" makes more sense. > > Why not engineer something to scale and last without knowing that you > will have to revisit it and quite possibly at the most inopportune > time, like when you just spent a good deal of money on an advertising > spot? > > Thanks, > Steve T > > On Wed, Jan 2, 2013 at 7:35 AM, Leandro Dardini <[email protected]> wrote: >> I don't know how many I/O can be achieved on a modern hardware, but I don't >> think 60 concurrent calls will be a problem. 60 calls are just 4 Mbit/s of >> data. However can be a good idea to start loading a server and be prepared >> to share the load on another server. >> >> Leandro >> >> >> 2013/1/2 Steve Totaro <[email protected]> >>> >>> Top post for the New Year. >>> >>> Yes, if you might scale up to 60 or more simultaneous calls, >>> definitely look at OrecX or RTPTap because you will run into I/O >>> issues. Not sure what current hardware can accommodate but it is best >>> not to find out. >>> >>> Considering the very low cost of hardware these days compared with the >>> cost of possible downtime, poor audio, lost recordings or whatever >>> else you can assign a monetary value, I always suggest a separate >>> machine for "Passive" recording when dealing with more than a handful >>> of simultaneous calls. >>> >>> Thanks, >>> Steve Totaro >>> >>> On Wed, Jan 2, 2013 at 6:18 AM, Lenz Emilitri <[email protected]> >>> wrote: >>> > With just one PRI card this should not be an issue, but for larger >>> > systems >>> > you may consider using something like Oreka to offload the I/O from the >>> > Asterisk server.... >>> > l. >>> > >>> > >>> > 2012/12/31 Vinod Nadiadwala <[email protected]> >>> >> >>> >> Hi, >>> >> >>> >> I am new to asterisk, i want to know that is it possible to use >>> >> asterisk >>> >> for build voice recording system. >>> >> >>> >> Scenario : >>> >> ISDN PRI line (30 line) >>> >> I want every incoming & outgoing call has to recorded, but without >>> >> manual >>> >> action. system has to auto receive the call. >>> >> >>> >> Please suggest, how should i start and with which hardware / cards it >>> >> is >>> >> possible. >>> >> >>> >> >>> >> >>> >> >>> >> -- >>> >> _____________________________________________________________________ >>> >> -- 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 >>> > >>> > >>> > >>> > >>> > -- >>> > Loway - home of QueueMetrics - http://queuemetrics.com >>> > Test-drive WombatDialer beta @ http://wombatdialer.com >>> > >>> > -- >>> > _____________________________________________________________________ >>> > -- 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 -- _____________________________________________________________________ -- 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
