<br><br><div class="gmail_quote">On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen <span dir="ltr"><<a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><div></div><div class="h5">On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:<br>
> On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen <<a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a>>wrote:<br>
><br>
> > On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:<br>
> > > On Sun, Sep 6, 2009 at 10:47 PM, Research <<a href="mailto:research@businesstz.com">research@businesstz.com</a>><br>
> > wrote:<br>
> > ><br>
> > > > Hello team;<br>
> > > > While am aware and active user of astersk monitor function for<br>
> > recording, i<br>
> > > > would like to know if i can use asterisk as a pure recording<br>
> > server(like<br>
> > > > nice or witness) for some other PABX's extensions (both inbound,<br>
> > outbound<br>
> > > > and internal).<br>
> > > ><br>
> > > > Setup<br>
> > > > PSTN---Legacy PABX(with analogy n digital extensions)---<br>
> > asterisk(record<br>
> > > > Legacy PABX extensions.)<br>
> > > ><br>
> > > > Sam<br>
> > > ><br>
> > > ><br>
> > > Is there any SIP or other VoIP in the mix? If so, you should take a look<br>
> > at<br>
> > > OrecX.<br>
> > > <a href="http://oreka.sourceforge.net" target="_blank">http://oreka.sourceforge.net</a> (Open Source)<br>
> > > They also have a paid version.<br>
> ><br>
> > Another method to do that is to make the Asterisk monitor output dummy<br>
> > SIP calls rather than sound files. Oreka/Orex can listen to those.<br>
> ><br>
> > Looking for volunteers to test that:<br>
> ><br>
> > <a href="http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/" target="_blank">http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/</a><br>
> > <a href="http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/" target="_blank">http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/</a><br>
> ><br>
> > <a href="http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample" target="_blank">http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample</a><br>
> ><br>
> > This allows recording non-VoIP links, VoIP links where tapping is not<br>
> > convinient, or more selective recording of VoIP calls.<br>
> ><br>
><br>
> Is this similar or the same as the portion of my post that you snipped?<br>
<br>
</div></div>Different in many ways, which is why I snipped it.<br>
<div class="im"><br>
><br>
> "Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but<br>
> minus the VoIP."<br>
<br>
</div>(Actually: recorded calls are sent as RTP streams to the Orex/Oreka<br>
server)<br>
<br>
This records outside of Asterisk. Thus it lacks information available in<br>
Asterisk (who really called who). OTOH, it is Asterisk-specific.<br>
<br>
We actually considered implementing something similar to the Sangoma<br>
interface in our driver but realised that doing it in Asterisk would<br>
probably be more useful. The overheade seems reasonable.<br>
<div><div></div><div class="h5"><br clear="all"></div></div></blockquote></div><br>Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk.....<br>
<br>Your quote "This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls."<br><br>Isn't that more or less the same thing I said that you snipped, "Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP."<br>
<br>This isn't the biz list, nor the dev list. Snipping out the reference of Sangoma being able to do RTP tap and suggesting people use your experimental dev branch doesn't really help users very much.<br><br>I really enjoy your use of selective snipping, quoting, and taking things out of context to manipulate threads. You should be a reporter. Too bad it doesn't work on me and I will call you out on it.<br>
<br>Please let us users know when your branch gets merged into a "Stable Release"<br><br>-- <br>Senior Systems and Network Administrator<br>Triple Canopy, Inc.,<br>2250 Corporate Park Drive, Suite 300<br>ph. +1.703.673.5191<br>
mob.+1.240.938.1212<br>FAX.+1.703.673.1279<br><a href="mailto:steve.totaro@triplecanopy.com">steve.totaro@triplecanopy.com</a><br>