[asterisk-bugs] [Asterisk 0015288]: Aborted trunk call into FXO keeps destination FXS port ringing

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15288 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15288
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 199298 
Request Review:              
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Date Submitted:             2009-06-08 05:16 CDT
Last Modified:              2009-07-24 04:52 CDT
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Summary:                    Aborted trunk call into FXO keeps destination FXS
port ringing
Description: 
Confirmed with a TDM800P with 1x 1port FX0 module and 1x 4port FXS module.

The only way to stop the ringing extension, was to pick it up.

The senario was:
  FX0 connected to outside line.
  FXS connected to analog phone.
  ring home line (from cellphone).
  wait for internal FXS connected phone to ring 1 or 2 rings. Hangup call
from mobile. 
  FXS connected phone keeps ringing forever.
  To stop the ringing, pickup ringing handset.

Deployed 1.6.1.0 branch, with same configs, no problems now.
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---------------------------------------------------------------------- 
 (0108159) alecdavis (reporter) - 2009-07-24 04:52
 https://issues.asterisk.org/view.php?id=15288#c108159 
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DAHDI Version: 2.2.0 Echo Canceller: MG2
Asterisk SVN-trunk-r208504
Hardware.
 TDM800P with 1x 4 PORT FXS module, 1x 1 port FXO module.

Tried latest trunk tonight, just to check whether status has changed.
Still the same.

Calling in on to FXO port to a TDM800P port keeps extensions ringing after
caller aborts call before being answered.

Only way to stop ringing extension is to pick it up and hangup.

console output below with FXS ports being rung:
Note: the caller has hungup 30 seconds ago!.
    -- DAHDI/2-1 is ringing
    -- DAHDI/3-1 is ringing
    -- DAHDI/4-1 is ringing
    -- DAHDI/2-1 is ringing
    -- DAHDI/3-1 is ringing
    -- DAHDI/4-1 is ringing
    -- DAHDI/4-1 answered DAHDI/5-1
    -- Hanging up on 'DAHDI/3-1'
    -- Hungup 'DAHDI/3-1'
    -- Hanging up on 'DAHDI/2-1'
    -- Hungup 'DAHDI/2-1'
    -- Native bridging DAHDI/5-1 and DAHDI/4-1
    -- Hanging up on 'DAHDI/4-1'
    -- Hungup 'DAHDI/4-1'
  == Spawn extension (incoming, s, 1) exited non-zero on 'DAHDI/5-1'
    -- Hanging up on 'DAHDI/5-1'
    -- Hungup 'DAHDI/5-1'

/etc/dahdi/system.conf
loadzone = nz
defaultzone=nz
fxols=1-4
fxsls=5

/etc/asterisk/chan_dahdi.conf
ringtimeout=2000

context=phones
signalling=fxo_ls
echocancel=yes
echotraining=yes
;echocancel=256
echocancelwhenbridged=yes
busydetect=yes
busycount=4
answeronpolarityswitch=no
hanguponpolarityswitch=no
group=1
callerid=Garage <81>
channel=>1
callerid=Study <85>
channel=>2
callerid=Master Bedroom <82>
channel=>3
callerid=Kitchen <89>

context=incoming
signalling=fxs_ls
rxgain=1.0
txgain=1.0
group=0
busydetect=no
busycount=4
answeronpolarityswitch=no
hanguponpolarityswitch=no
callerid=asreceived
channel=>5

/etc/asterisk/extensions.conf
[incoming]
exten => s,1,Dial(DAHDI/2&DAHDI/3&DAHDI/4)

===============================================================================

simplified dialplan using just a SIP phone:

[incoming]
exten => s,1,Dial(SIP/bt100black)

console output: SIP phone keeps ringing after aborted inbound call.

   -- Starting simple switch on 'DAHDI/5-1'
[Jul 24 21:40:04] NOTICE[7614]: chan_dahdi.c:2642
my_handle_notify_message: MWI: Channel 5 message waiting, mailbox
    -- Executing [s at incoming:1] Dial("DAHDI/5-1", "SIP/bt100black") in new
stack
    -- Called bt100black
    -- SIP/bt100black-09e1cee8 is ringing
    -- SIP/bt100black-09e1cee8 answered DAHDI/5-1
  == Spawn extension (incoming, s, 1) exited non-zero on 'DAHDI/5-1'
    -- Hanging up on 'DAHDI/5-1'
    -- Hungup 'DAHDI/5-1'

same configuration has been working for weeks with an earlier version of
Asterisk SVN 1.6.1 branch

Just updated to SVN-branch-1.6.1-r208503 and it works as expected, with
exactly same configurations as I'm reporting with. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-24 04:52 alecdavis      Note Added: 0108159                          
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