Thanks for the script, I did have one question. Your script mentions using a column labeled "timestamp" using UNIX_TIMESTAMP. I'm assuming of course this is, well, unix time :-) I couldn't find anywhere in the acc module docs to output "CDR's" in unixtime, did you have something attached to SER to convert it? Thanks!
Matt
-----Original Message----- From: Darren Nay [mailto:dnay@IONOSPHERE.net] Sent: Thursday, December 16, 2004 11:59 AM To: serusers@lists.iptel.org Subject: RE: [Serusers] Does SER do that?
Sure, I am happy to share it.
Keep in mind. This script assumes that you are using a phone number (DID) username scheme. So if your usernames aren't in a standard DID format of 1XXXXXXXXXX or +1XXXXXXXXXX then you might have to tweak it some to make it work correctly for you.
Give it a try and give me some feedback. Thanks!
Darren Nay
-----Original Message----- From: Matt Schulte [mailto:mschulte@netlogic.net] Sent: Thursday, December 16, 2004 11:58 AM To: Darren Nay; Alberto Martínez; serusers@lists.iptel.org Subject: RE: [Serusers] Does SER do that?
Nice! Care to share the app? :-) heehee
-----Original Message----- From: Darren Nay [mailto:dnay@IONOSPHERE.net] Sent: Thursday, December 16, 2004 10:39 AM To: Matt Schulte; Alberto Martínez; serusers@lists.iptel.org Subject: RE: [Serusers] Does SER do that?
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@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@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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