Hi,
I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. [snip]
However in the opposite direction, Norstar -> SIP, the problem begins, thanks to no disconnect supervision/provision on the FXS port from the Norstar, so the X100P doesn't know the Norstar caller has hangup.
A thought that occurred was perhaps Asterisk could do some sort of "soft" disconnect supervision? So when the FXO card has seized the line, but there is no transmit/receive audio (or no major variation from the standard background radiation) then after a safe timeout of say a minute or two, it could disconnect the line? I'd be interested to hear others views on this, would it work (or does it already exist?). [snip] Cheers, Chris Lee
No, though there has been some discussion about similar problems with disconnect supervision with SIP. Of course, the SIP disconnect supervision is packet-based (where there has been no reinvite) and your problem is audio-based; it's the same method, though - check to see if there is activity on the session in the TX or RX direction, and if no activity crossing threshold Y for more than N seconds, hang up.
See http://bugs.digium.com/bug_view_page.php?bug_id=0000207 for more discussion.
I think this is pretty reasonable, actually. We have such a system in voicemail - if there is silence at the end of a message, the system hangs up and rewinds the VM recording to the last spike in audio energy. You will get people yelping at you about how this breaks conference calling, or maybe it breaks prank calling. My answer to this is that it should be a selectable switch/values on the Dial. app_dial sure is getting crowded with ideas, but is still pretty lean on features.
Anyone feel like a) programming this or b) funding it?
JT _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
