Check out Oreka at sourceforge, too.(aka OrkAudio)<br><br><div><span class="gmail_quote">On 2/15/07, <b class="gmail_sendername">Kristian Kielhofner</b> <<a href="mailto:kristian.kielhofner@gmail.com">kristian.kielhofner@gmail.com
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">On 2/15/07, Cory Andrews <<a href="mailto:cory@voipsupply.com">cory@voipsupply.com
</a>> wrote:<br>> Apologies in advance as this is not directly Asterisk related, however I<br>> thought I might be able to leverage the experience of particiapants on<br>> this listserv for some advice.<br>>
<br>> I have a client who is utilizing Talkswith PBX appliances, which have no<br>> native call monitoring/call recording capabilities. They are looking<br>> for an additional application, service or appliance that can sit on the
<br>> LAN, and allow an administrator to monitor or recording inbound/outbound<br>> calls. If anyone is aware of a mechanism or solution that would provide<br>> this capability, please shoot me an email.<br>>
<br>> Thanks<br>><br>> Cory Andrews<br><br>Cory,<br><br> From their website it appears they are using SIP. With any luck it<br>will be SIP + ulaw (without re-invites). If so, do this:<br><br>1) Get a decent managed switch that can setup monitor ports.
<br>Configure one port to monitor the port connected to the Talkswitch.<br><br>2) Get a decent dual-homed machine.<br><br>3) Connect one interface of the dual-homed machine to the monitor<br>port. Running Linux, do an ifconfig up [interface name] (no IP
<br>address). Configure the other interface to connect to a network for<br>management, copying files, etc.<br><br>4) Start up tcpdump on the interface, writing to a file.<br><br>5) Use something like Cain + Abel to read the RTP and dump the audio to a file.
<br><br>6) Convert files to desired format using sox.<br><br> The only step I left out was "Profit!". Seriously though, this<br>depends on a few key assumptions about the Talkswitch:<br><br>1) That it is standard SIP.
<br><br>2) It uses ulaw.<br><br>3) It doesn't do re-invites.<br><br> Not any one of these is a show stopper for this type of sollution,<br>but any one of them (or all of them) could make life a bit harder for<br>you...
<br><br> Good luck!<br><br>--<br>Kristian Kielhofner<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list
<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br>