[asterisk-bugs] [DAHDI-linux 0015931]: Dial() command do not detect hangup during extension call

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Sep 22 09:52:38 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15931 
====================================================================== 
Reported By:                Francis Brosnan
Assigned To:                
====================================================================== 
Project:                    DAHDI-linux
Issue ID:                   15931
Category:                   wctdm
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
JIRA:                        
====================================================================== 
Date Submitted:             2009-09-22 09:01 CDT
Last Modified:              2009-09-22 09:52 CDT
====================================================================== 
Summary:                    Dial() command do not detect hangup during extension
call
Description: 
If we receive a call from the analogic device (TDM410P), it gets answered
correctly, but if the caller hangups during Dial() command is ringing to
default extension, this hangup is not detect and local extension keeps on
ringing until someone pick up the phone (or pick the extension) or timeout
is reached.


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

---------------------------------------------------------------------- 
 (0111190) Francis Brosnan (reporter) - 2009-09-22 09:52
 https://issues.asterisk.org/view.php?id=15931#c111190 
---------------------------------------------------------------------- 
>> Why do you believe this is an Asterisk, Dial application problem, rather
than 
>> a problem with Dahdi, or maybe chan_dahdi? 

Hi David. I believe it is a problem with Dial application because it
continues dialing even having the line hanged up. Maybe it is not Dial()
cmd itself but how our hardware is detected. 

>> The console output should have debugging enabled per logger.conf along
with 
>> 'core set debug 10' 

Hi. We are using zaptel (1.4.12.1), libpri (1.4.10.1). Not dahdi. Our
/etc/asterisk/zapata.conf content is:
[channels]
context=incoming
language=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes

signalling=fxs_ks

; we have checked the following two options but no success
; hanguponpolarityswitch=yes
; busydetect=yes
------------

channel => 1

Our /etc/zaptel.conf configuration is:

# Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) 
fxsks=1
# channel 2, WCTDM/0/1, no module.
# channel 3, WCTDM/0/2, no module.
# channel 4, WCTDM/0/3, no module.
loadzone        = es
defaultzone     = es

Setting "core set debug 10" produces the following log when a call is
received:

    -- Starting simple switch on 'Zap/1-1'
    -- Executing [s at incoming:1] GotoIfTime("Zap/1-1",
"09:00-14:00|mon-fri|*|*?label_abierto") in new stack
    -- Executing [s at incoming:2] GotoIfTime("Zap/1-1",
"15:00-18:00|mon-fri|*|*?label_abierto") in new stack
    -- Goto (incoming,s,4)
    -- Executing [s at incoming:4] Answer("Zap/1-1", "") in new stack
    -- Executing [s at incoming:5] Playback("Zap/1-1",
""../custom-sounds/bienvenido-aspl"") in new stack
    -- <Zap/1-1> Playing '../custom-sounds/bienvenido-aspl' (language
'es')
    -- Executing [s at incoming:6] Dial("Zap/1-1", "SIP/sip3&SIP/sip4|30|m")
in new stack
    -- Called sip3
    -- Called sip4
    -- Started music on hold, class 'default', on Zap/1-1
    -- SIP/sip3-09c3b068 is ringing
    -- SIP/sip4-09c4b5a0 is ringing

...at this point, if I hangup (I'm calling from a mobile), no message is
found and sip3/sip4 extensions keep on ringing. At this point, after
waiting a few seconds, I pickup the ringing (phantom) call from my sip
extension but nothing is found. Here is the log produced when I pickup:

    -- Executing [40 at phones:1] PickUp2("SIP/francissip-09c452e8", "SIP")
in new stack
       > find_matching_channel: pattern='SIP' option='' state=5
       > find_matching_channel: trying channel='SIP/francissip-09c452e8'
state=4 pattern='SIP'
       > find_matching_channel: trying channel='SIP/sip4-09c4b5a0' state=5
pattern='SIP'
       > find_matching_channel: found channel='SIP/sip4-09c4b5a0'
    -- Channel SIP/francissip-09c452e8 picked up ringing channel
SIP/sip4-09c4b5a0
    -- Executing [40 at phones:2] Hangup("SIP/francissip-09c452e8<MASQ>", "")
in new stack
  == Spawn extension (phones, 40, 2) exited non-zero on
'SIP/francissip-09c452e8<MASQ>'
    -- SIP/francissip-09c452e8 answered Zap/1-1
    -- Stopped music on hold on Zap/1-1
Really destroying SIP dialog
'70d10db419e391e0037960e63147589d at 192.168.0.233' Method: INVITE
Really destroying SIP dialog
'554526ca1f9dc7c674be5ec5529d81d3 at 192.168.0.233' Method: INVITE
  == Spawn extension (incoming, s, 6) exited non-zero on 'Zap/1-1'
    -- Hungup 'Zap/1-1'

If I execute "core show channels" after hang up, it is showed the
following:
sturgeon*CLI> 
    -- Starting simple switch on 'Zap/1-1'
    -- Executing [s at incoming:1] GotoIfTime("Zap/1-1",
"09:00-14:00|mon-fri|*|*?label_abierto") in new stack
    -- Executing [s at incoming:2] GotoIfTime("Zap/1-1",
"15:00-18:00|mon-fri|*|*?label_abierto") in new stack
    -- Goto (incoming,s,4)
    -- Executing [s at incoming:4] Answer("Zap/1-1", "") in new stack
    -- Executing [s at incoming:5] Playback("Zap/1-1",
""../custom-sounds/bienvenido-aspl"") in new stack
    -- <Zap/1-1> Playing '../custom-sounds/bienvenido-aspl' (language
'es')
    -- Executing [s at incoming:6] Dial("Zap/1-1", "SIP/sip3&SIP/sip4|30|m")
in new stack
    -- Called sip3
    -- Called sip4
    -- Started music on hold, class 'default', on Zap/1-1
    -- SIP/sip3-09c3b068 is ringing
    -- SIP/sip4-09c4b5a0 is ringing
<...hangup call from my mobile and wait a few seconds...the run: >
sturgeon*CLI> core show channels
Channel              Location             State   Application(Data)       
     
SIP/sip4-09c4b5a0    s at phones:1           Ringing AppDial((Outgoing Line))
     
SIP/sip3-09c3b068    s at phones:1           Ringing AppDial((Outgoing Line))
     
Zap/1-1              s at incoming:6         Up     
Dial(SIP/sip3&SIP/sip4|30|m)  
3 active channels
1 active call

After looking at this data, I've checked to also hangup during playback()
and the problem persists so it looks like is not a problem with Dial()...

Let me know if I can provide you with more details...Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-22 09:52 Francis BrosnanNote Added: 0111190                          
======================================================================




More information about the asterisk-bugs mailing list