[asterisk-users] Using asterisk as the recording server

Tue Sep 8 01:43:31 CDT 2009

On Mon, Sep 07, 2009 at 01:47:57PM -0400, Steve Totaro wrote:
> 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...

Which means you have to keep a separate DB of that (I know such DB
exists: the CDR) and get that data from it. Extra work to do. Some
people prefer to avoid it.

> 
> 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.

VoIP mirroring implies you have control over the network infrastructure.
What if you install the PBX in a hostile network where the network
administrator doesn't like you sniffing other network traffic?

Not to mention that it is extra setup.

So we add a different option. One that depends on Asterisk sending the
relevant data, and uses the existing monitoring infrastructure in
Asterisk: simply use Monitor and StopMonitor to enable/disable
monitoring. This is something Asterisk admins should be familiar with.

> > 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.

My target audince in posts to asterisk-users is (surpirse-surpirse) the
readers of asterisk-users. I generally do expect them to follow the
list[1].

> 
> 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.

Right. He basically wrote in his document: "using BRI in the US is
possible but it requires *me* installing it". I don't like keeping
things secret. If it's possible, I want to document it. This means it
also works with our hardware but also other BRI adapters from other
vendors, including even cheap HFC-S cards.

> 
> 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.

I gather you don't volunteer to test that patch.

(And no: no chance for it getting merged into 1.4. Unless something very
unlikely happens, it won't even get into 1.6.2, as it is feature-frozen
already. I hope it will make it into 1.6.3. But there's lots of work on
it)

[1] That is: I don't expect all readers to read ever message. But
normally readers give up on a thread at some early point in it and hence
the chances of missing a parent message and reading just the child are
rather small.

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir