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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Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Great! I have installed with mysql support and authentification works well. Now I am looking how to redirect connectios to the SIP provider server. Anyone can help me? I want all connections to be redirected to the same sip telephony provider server. SER will only get the control of the users allowing or not to pass through and get calling time control so we could end a call if it surpases the max time or something like this.
DN> Yes, this can be done to some degree.
DN> I've written a perl script that queries the "acc" table and keeps track of DN> current calls. The only issue is that occasionally if a call is not broken DN> down correctly (ie. SER did not receive the BYE) then you may have some DN> calls reported to stay up when actually they are not.
DN> However, the best way to keep track of concurrent calls is to use DN> mediaproxy. It always knows exactly how many calls are up because it DN> carries the audio for each call. Much more accurate than using sers "acc" DN> table.
DN> That being said, we still use the "acc" table method because we prefer that DN> the audio streams not ride our network. It saves cost on bandwidth and DN> we've found that the voice quality is better directly from IAD to PSTN/IAD DN> and not relayed through an RTP proxy.
DN> 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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-
Alberto Martínez -
Darren Nay