The ring splash is a long standing feature of call forwarding. Of course somewhere in the Asterisk code a change could be made to extend the time required to detect a valid ring.
But, how about just unplugging the pots lines from the PBX with a quick restore ability? Unplug lines at the NID, or open bridging clips or whatever applies. Cary Fitch -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, May 26, 2010 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] "ring splash" Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a "ghost" call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said "oh yeah - that is a "ring splash" that lets the customer know that a call was forwarded". They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a "ring splash" and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
