yep..that would be what i said, using the nifty slang my "peeps" use in the datacenters....
I just wanted to be "cool" like them...*hangs head*... great...now I gotta transfer to another school... LOL, have a good one mate! On Tue, Feb 8, 2011 at 7:23 AM, <[email protected]> wrote: > Yes. The technology need to be used on LAN switches is "port mirroring" or > "line tapping" > > > > > -----Original Message----- > From: "Sherwood McGowan" <[email protected]> > Sent: Tuesday, February 8, 2011 7:34am > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > [email protected]> > Subject: Re: [asterisk-users] Call Recording audio file quality query > > On Tue, Feb 8, 2011 at 6:01 AM, <[email protected]> wrote: > >> But if you are getting calls all the way on VoIP then you can have calls >> in HD audio using HD audio codec on all locations (Server and Client). In >> that case you either need use some available 3rd party solution which uses >> packet capturing to trace the calls and record call using packet capture and >> assembling regardless of server as asterisk still will not be able to record >> call in HD but some other switches like FreeSWITCH can do it or you need to >> write your own app like it. >> >> > > It's not difficult at all to perform what you're referring to..If you have > the hardware... > > A simple way is to have a port on your main network switch/router that will > "firehose" the traffic the device interacts with In case someone reading > this doesn't know, I'm talking about having a port that just makes a copy of > EVERY PACKET that the device "sees" and sends those copies out over the port > that you've set up for the purpose..It just GUSHES data over that > port...like a firehose just gushes out all the water it possibly can... LOL > > Anyway, once your data is being mirrored over that firehose, send it to a > dedicated "recording" server...all it has to do is find the signaling > packets for each call and then just dump the "payload" from the RTP. It'll > come out exactly as it was transported within RTP...in the codec the call > set up.... > > I may be wrong, but I'm fairly sure that Asterisk can write a filetype for > almost any of it's codecs...I know it can READ audio files that are encoded > in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... > > If the "DECoding" portion is there, there's almost GOT to be the "enCOding" > functionality... > > > > -- > _____________________________________________________________________ > -- 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 >
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