[asterisk-users] Using asterisk as the recording server
Tzafrir Cohen
tzafrir.cohen at xorcom.com
Mon Sep 7 09:09:02 CDT 2009
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.
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.
>
> 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.
--
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
More information about the asterisk-users
mailing list