Re: [asterisk-users] Call Recording - Asterisk

2008-12-09 Thread Chris Rowson


 
  I wanted to setup Oreka to monitor calls on a trixbox box I have
  setup. Oreka doesn't seem to be catching all of the calls
  though I have port mirroring setup on the port that trixbox is
  connected to, mirrored to the port Oreka is connected to.
 
  I have read that Asterisk doesn't work as a SIP Proxy, so I
  wondered if this meant that some phones, after checking in with
  Asterisk would simply communicate via RTP between each other,
  without going media transport going through trixbox itself? If
  this is the case then I guess I'd need to mirror the full VoIP
  VLAN to the Oreka port wouldn't I? Or is there another reason that
  I'm missing here?
 

 Chris,

 Make sure that all of your SIP clients are set to canreinvite=no in
 sip.conf.  The default is canreinvite=yes, which allows RTP to
 bypass Asterisk.  Certain things (codec translation, playback of audio
 files, etc.) require Asterisk to be in the RTP path, which may explain
 why you're recording some of the calls.

 If you're still missing calls, make sure Oreka is configured properly in
 config.xml.  In particular, the AllowedIpRanges and
 BlockedIpRanges settings provide IP address filtering at the Oreka
 level.  In general, I've had to configure these to prevent getting two
 recordings of each call (since Asterisk acts as a B2BUA) but your
 configuration may be too strict.

 Running tcpdump/Wireshark on the Oreka server will let you see exactly
 what's being mirrored.  There is even a setting in Oreka named
 PcapFile that will let you playback the packet capture file over and
 over until you're satisfied with your configuration.

 Regards,

 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer


Matthew,

Thank you so much for your advice. It's really appreciated - I'll go through
it and see where I get.

Thanks again

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Recording - Asterisk

2008-12-08 Thread Chris Rowson

 Hello folks,

 I wanted to setup Oreka to monitor calls on a trixbox box I have setup.
 Oreka doesn't seem to be catching all of the calls though I have port
 mirroring setup on the port that trixbox is connected to, mirrored to the
 port Oreka is connected to.

 I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if
 this meant that some phones, after checking in with Asterisk would simply
 communicate via RTP between each other, without going media transport going
 through trixbox itself? If this is the case then I guess I'd need to mirror
 the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason
 that I'm missing here?

 Just trying to get this sussed out in my head!

 Thanks for your time.

 Chris


Hi again, didn't get a reply to this one. I'm a bit stumped so I thought I'd
try the list one more time to see if anyone has an answer.

If not, thanks for reading anyway!

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Recording - Asterisk

2008-12-08 Thread Matthew J. Roth
Chris Rowson wrote:

 I wanted to setup Oreka to monitor calls on a trixbox box I have
 setup. Oreka doesn't seem to be catching all of the calls
 though I have port mirroring setup on the port that trixbox is
 connected to, mirrored to the port Oreka is connected to.

 I have read that Asterisk doesn't work as a SIP Proxy, so I
 wondered if this meant that some phones, after checking in with
 Asterisk would simply communicate via RTP between each other,
 without going media transport going through trixbox itself? If
 this is the case then I guess I'd need to mirror the full VoIP
 VLAN to the Oreka port wouldn't I? Or is there another reason that
 I'm missing here?


Chris,

Make sure that all of your SIP clients are set to canreinvite=no in 
sip.conf.  The default is canreinvite=yes, which allows RTP to 
bypass Asterisk.  Certain things (codec translation, playback of audio 
files, etc.) require Asterisk to be in the RTP path, which may explain 
why you're recording some of the calls.

If you're still missing calls, make sure Oreka is configured properly in 
config.xml.  In particular, the AllowedIpRanges and 
BlockedIpRanges settings provide IP address filtering at the Oreka 
level.  In general, I've had to configure these to prevent getting two 
recordings of each call (since Asterisk acts as a B2BUA) but your 
configuration may be too strict.

Running tcpdump/Wireshark on the Oreka server will let you see exactly 
what's being mirrored.  There is even a setting in Oreka named 
PcapFile that will let you playback the packet capture file over and 
over until you're satisfied with your configuration.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] Call Recording - Asterisk

2008-12-06 Thread Chris Rowson
Hello folks,

I wanted to setup Oreka to monitor calls on a trixbox box I have setup.
Oreka doesn't seem to be catching all of the calls though I have port
mirroring setup on the port that trixbox is connected to, mirrored to the
port Oreka is connected to.

I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this
meant that some phones, after checking in with Asterisk would simply
communicate via RTP between each other, without going media transport going
through trixbox itself? If this is the case then I guess I'd need to mirror
the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason
that I'm missing here?

Just trying to get this sussed out in my head!

Thanks for your time.

Chris
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users