[Asterisk-Dev] Asterisk X100p can not hangup

[dev2003 at mail.ustc.edu.cn]

Zaptel zapata asterisk from cvs

use x100p card
When pstn fisrt hangup, asterisk can not find that the phone has hungup.


Asterisk Ready.
*CLI>     -- Starting simple switch on 'Zap/1-1'
Nov  6 16:42:07 NOTICE[4868]: chan_zap.c:5430 ss_thread: Got event 2
(Ring/Answered)...
Nov  6 16:42:09 ERROR[4868]: callerid.c:261 callerid_feed: fsk_serie made
mylen
< 0 (-16)
Nov  6 16:42:09 WARNING[4868]: chan_zap.c:5459 ss_thread: CallerID feed
failed:
Success
Nov  6 16:42:09 WARNING[4868]: chan_zap.c:5501 ss_thread: CallerID
returned with
error on channel 'Zap/1-1'
    -- Executing Dial("Zap/1-1", "SIP/24800027 at sip_proxy-out|20") in new
stack
    -- Called 24800027 at sip_proxy-out
    -- SIP/sip_proxy-out-2fc3 is ringing
    -- SIP/sip_proxy-out-2fc3 answered Zap/1-1

when I hangup the phone ,there is no respond.
How to do?

from google ,

http://www.marko.net/asterisk/archives/0208/0329.html



This has been covered in many other messages, but in order for the X100P

to detect hangup, you must have disconnect supervision on the phone line.

If the line you are providing is from another PBX, then it is highly

unlikely that it supplies the disconnect supervision. Most telephone

switches do support disconnect supervision, but it's not always on by

default. You can tell by using a lighted keypad which receives its power

only from the phone line, and then calling it and hanging up on it. If

the lighted keypad blinks off for a moment then your line has the

disconnect supervision, otherwise it doesn't.

http://www.marko.net/asterisk/archives/0206/0419.html
Modify chan_zap.c ???
How to modify code to find pstn first hangup?
thank a lot.




zaptel.conf

fxsks=1
# X100P

zapata.conf

signalling=fxs_ks

;context=from-sip
context =default
channel => 1