yep..that would be what i said, using the nifty slang my "peeps" use in the datacenters....<br><br>I just wanted to be "cool" like them...*hangs head*...<br>great...now I gotta transfer to another school...<br>
<br>LOL, have a good one mate! <br><br><div class="gmail_quote">On Tue, Feb 8, 2011 at 7:23 AM, <span dir="ltr"><<a href="mailto:faisal@vopium.com">faisal@vopium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<font size="2" face="verdana"><div>Yes. The technology need to be used on LAN switches is "port mirroring" or "line tapping"<div class="im"><br><br><br><br>-----Original Message-----<br>From: "Sherwood McGowan" <<a href="mailto:sherwood.mcgowan@gmail.com" target="_blank">sherwood.mcgowan@gmail.com</a>><br>
Sent: Tuesday, February 8, 2011 7:34am<br>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
Subject: Re: [asterisk-users] Call Recording audio file quality query<br><br>
</div><div><div></div><div class="h5"><div class="gmail_quote">On Tue, Feb 8, 2011 at 6:01 AM, <span dir="ltr"><<a href="mailto:faisal@vopium.com" target="_blank">faisal@vopium.com</a>></span> wrote:<br>
<blockquote style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;" class="gmail_quote"><font size="2" face="verdana">
<div>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.
<div>
<div></div>
<div><br></div></div></div></font></blockquote><br></div><br>It's not difficult at all to perform what you're referring to..If you have the hardware...<br><br>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<br>
<br>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....<br>
<br>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...<br>
<br>If the "DECoding" portion is there, there's almost GOT to be the "enCOding" functionality...<br><br><br></div></div></div></font><br>--<br>
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