>>>>> "FM" == Fabio Moretti <fmore...@tecytal.com> writes:
FM> when a call enter, asterisk sense it and store its values (callerid, FM> date and time, etc) somewhere, but nothing more, people will answer FM> using the old analog phone. The goal is to have a log of the inbound FM> calls without touching the old analog system (it's shared between FM> different subjects). IIUC, the pots line has both some number of analog phones a/o fax machines on it, plus a fxo->sip gateway, yes? You can route the sip portion to asterisk and have the dialplan log everything but never answer. You may want to call the Ringing dialplan application, but even that may not be required. OTOH, calling Ringing should prevent the gateway from assuming that the asterisk machine never saw the INVITE. Eventually, when the other extension answers, the fxo->sip gateway will cancel the sip call just like it would if the caller hung up. (There is a possibility that any given gateway may not cancel the sip call until the analog call is completed; you need to test.) -JimC -- James Cloos <cl...@jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
