New subject: Re[2]: [Serusers] Does SER do that?

17 Dec 2004

Yes, this can be done to some degree.

I've written a perl script that queries the "acc" table and keeps track of
current calls.  The only issue is that occasionally if a call is not broken
down correctly (ie. SER did not receive the BYE) then you may have some
calls reported to stay up when actually they are not. 

However, the best way to keep track of concurrent calls is to use
mediaproxy.  It always knows exactly how many calls are up because it
carries the audio for each call.  Much more accurate than using sers "acc"
table.

That being said, we still use the "acc" table method because we prefer that
the audio streams not ride our network.  It saves cost on bandwidth and
we've found that the voice quality is better directly from IAD to PSTN/IAD
and not relayed through an RTP proxy.

Darren

...
  -----Original Message-----
 From: Matt Schulte [mailto:mschulte@netlogic.net]
 Sent: Thursday, December 16, 2004 11:38 AM
 To: Alberto Martínez; serusers(a)lists.iptel.org
 Subject: RE: [Serusers] Does SER do that?

 Yes, yes, and maybe.

  I  would like to know if SER can notify by anyway
to other application  of  the  start  and  end  of
  VoIP transmissions in order to maintain a control
in real-time of calls  made by the users.

 This one maybe a little more tricky, I'm looking for such an app myself. I
 suppose this could be done via the acc module.

 -----Original Message-----
 From: Alberto Martínez [mailto:amartinez@astrasoft.es]
 Sent: Thursday, December 16, 2004 9:21 AM
 To: serusers(a)lists.iptel.org
 Subject: [Serusers] Does SER do that?

 -----BEGIN PGP SIGNED MESSAGE-----
 Hash: MD5

 Hello,

 I  am  new  in  SER.  I  would  like  to  ask you if SER is able to do
 somethings I am looking for. I would like it to check if the users who try
 to  connect  with  it by SIP are correct, checking the login info and, if
 they are, redirect the connection to the VoIP provider. If the user  who
 is  trying  to  connect  is  not  set  up in SER it must be rejected.

 I  would like to know if SER can notify by anyway to other application of
 the  start  and  end  of VoIP transmissions in order to maintain a control
 in real-time of calls made by the users.

 Thank you.

 Best regards,
 Alberto

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