[asterisk-bugs] [Asterisk 0011993]: asterisk crashes when trying to make a call from SIP endpoint to h323 endpoint registered to gnugk

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11993 
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Reported By:                tsgan
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11993
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.17 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             02-14-2008 03:44 CST
Last Modified:              02-14-2008 04:56 CST
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Summary:                    asterisk crashes when trying to make a call from SIP
endpoint to h323 endpoint registered to gnugk
Description: 
asterisk crashes and segmentation faults with core dump when trying to make
a call from SIP endpoint to h323 endpoint registered to gnugk.
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---------------------------------------------------------------------- 
 tsgan - 02-14-08 04:56  
---------------------------------------------------------------------- 
h323.conf

; The NuFone Network's
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddress=192.168.0.233
disallow=all
allow=gsm               ; Always allow GSM, it's cool :)
allow=ulaw              ; see doc/rtp-packetization for framing options
allow=alaw
allow=g723
allow=g729
allow=gsm

gatekeeper = 192.168.0.18

sip.conf

[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support.
(Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
relaxdtmf=yes                   ; Relax dtmf handling

[1100051]
type=friend
username=1100051
;secret=4321
;nat=yes
host=dynamic
context=default
;defaultip=192.168.0.120
canreinvite=no
callerid=1100051
mailbox=1100051 at local
disallow=all
allow=alaw
allow=ulaw
;allow=gsm
;allow=g729
;allow=g723

extention.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for
demo
IAXINFO=guest                                   ; IAXtel
username/password
TRUNK=Zap/g0                                    ; Trunk interface

TRUNKMSD=1                                      ; MSD digits to strip
(usually 1 or 0)


[default]
exten => s,1,Wait(2)
exten => s,2,NoOp
exten => s,3,NoOp(${CALLERID(all)})
exten => s,4,NoOp(${CALLERID(num)})
exten => s,5,Dial(SIP/804,45)
exten => s,6,Hangup

; used to record prompts
exten => 205,1,Wait(2)
exten => 205,2,Record(/tmp/greetings:alaw)
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/greetings)
exten => 205,5,Wait(2)
exten => 205,6,Hangup

exten => _001.,1,Dial(H323/${EXTEN})
exten => _10.,1,Dial(OOH323/${EXTEN})

exten => _5555.,1,Dial(OOH323/${EXTEN})
exten => _1111.,1,Dial(OOH323/${EXTEN})

exten => _005.,1,Dial(OOH323/${EXTEN})

exten => _11xxxxx,1,Dial(SIP/${EXTEN},3600,t) 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-14-08 04:56  tsgan          Note Added: 0082231                          
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