Erick Perez wrote:
PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13
Asterisk is being used as a meetme server for 8 more calls.
Everything works fine in terms of the asterisk/meetme. The issue
arises when the calls comes in via the ATA286 box and in any part of
the meeting the CALLER hangs up but the ata286 does not realize the
caller hung up so the channels remains open and everyone in the room
hears a "busy" signal. After 30 seconds the ATA286 hangs up (I guess
due to timeout) and then the tdm04b hungs the channel and then the
meetme room gets back to normal.
The ATA will be getting the hangup - it'll be what's generating the busy
tone you hear when the SIP session between the ATA and your VoIP
provider is terminated.
If you can get your provider to enable the P205 "Polarity Reversal"
setting on the ATA, the ATA will reverse the polarity of the voltage on
it's FXS port when and outgoing call is answered (outbound calls), and
when the remote end hangs up (for calls in either direction).
You'll then be able to set hanguponpolarityswitch=yes in zapata.conf,
and hangups should then be detected almost immediately (with luck,
before any tones are heard).
HTH,
Nic.
--
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
