[asterisk-users] Using asterisk as the recording server

[Steve Totaro]

On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com>wrote:

> On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
> > On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com
> >wrote:
> >
> > > On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
> > > > On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen <
> tzafrir.cohen at xorcom.com
> > > >wrote:
> > > >
> > > > > On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
> > > > > > On Sun, Sep 6, 2009 at 10:47 PM, Research <
> research at businesstz.com>
> > > > > wrote:
> > > > > >
> > > > > > > Hello team;
> > > > > > > While am aware and active user of astersk monitor function for
> > > > > recording, i
> > > > > > > would like to know if i can use asterisk as a pure recording
> > > > > server(like
> > > > > > > nice or witness) for some other PABX's extensions (both
> inbound,
> > > > > outbound
> > > > > > > and internal).
> > > > > > >
> > > > > > > Setup
> > > > > > > PSTN---Legacy PABX(with analogy n digital extensions)---
> > > > > asterisk(record
> > > > > > > Legacy PABX extensions.)
> > > > > > >
> > > > > > > Sam
> > > > > > >
> > > > > > >
> > > > > > Is there any SIP or other VoIP in the mix?  If so, you should
> take a
> > > look
> > > > > at
> > > > > > OrecX.
> > > > > > http://oreka.sourceforge.net (Open Source)
> > > > > > They also have a paid version.
> > > > >
> > > > > Another method to do that is to make the Asterisk monitor output
> dummy
> > > > > SIP calls rather than sound files. Oreka/Orex can listen to those.
> > > > >
> > > > > Looking for volunteers to test that:
> > > > >
> > > > >  http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/
> > > > >  http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/
> > > > >
> > > > >
> > >
> http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample
> > > > >
> > > > > This allows recording non-VoIP links, VoIP links where tapping is
> not
> > > > > convinient, or more selective recording of VoIP calls.
> > > > >
> > > >
> > > > Is this similar or the same as the portion of my post that you
> snipped?
> > >
> > > Different in many ways, which is why I snipped it.
> > >
> > > >
> > > > "Sangoma RTP Tap will allow you to record TDM calls, again using
> OrecX
> > > but
> > > > minus the VoIP."
> > >
> > > (Actually: recorded calls are sent as RTP streams to the Orex/Oreka
> > > server)
> > >
> > > This records outside of Asterisk. Thus it lacks information available
> in
> > > Asterisk (who really called who). OTOH, it is Asterisk-specific.
> > >
> > > We actually considered implementing something similar to the Sangoma
> > > interface in our driver but realised that doing it in Asterisk would
> > > probably be more useful. The overheade seems reasonable.
> > >
> > >
> > 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.....
> >
> > 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."
> >
> > 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."
>
> And what if the call does not go through a TDM card? And ore
> importantly: how can you tell who is the caller and who is the callee?
> The rtp-tap interface basically tells you that channel X had a call at
> time Y.
>
>
I am sure it is pretty trivial to figure out who channel X and Y are based
on the channel, time, CID, DID....  Just a wee bit of code...

If it does not go through a TDM card, and is VoIP, then port mirroring works
just fine.  Sipcallid is a very simple way to match callers to callees.


> If you control recording through the monitoring interface of Asterisk
> you can start and stop the recording when you need it. You can also
> provide better information aobut the call. But again, it means that this
> is part of Asterisk, and I figure Sangoma has quite a few non-Asterisk
> customers.
>
> Sounds neat, when will it be out of beta?


>
>
>
> > 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.
>
> My message was an explicit call for testers, if you haven't noticed :-)
>
> I snip content that is not relevant to my reply. Whoever reads this list
> already read about the Sangoma interface previously. I had nothing to
> say about it. It was not related to that new branch.
>
>
Not everyone who reads the list, reads all the posts, give me a break.  It
was related to the thread.

Your motives and alliances have and always will be for Xorcom and Digium.
That is the only reason why you "helped" me with that BRI install in the US,
so you could poke around and try to figure out how Marcin Pycko achieved
what you cannot.

I may check it out when it is part of a "stable backported to 1.4" release,
otherwise, I don't run beta in production.

Sometimes large sums of money rely on systems, as do much more valuable
human lives.

-- 
Senior Systems and Network Administrator
Triple Canopy, Inc.,
2250 Corporate Park Drive, Suite 300
ph.   +1.703.673.5191
mob.+1.240.938.1212
FAX.+1.703.673.1279
steve.totaro at triplecanopy.com
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