[asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Nov 13 12:24:48 CST 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11233 
====================================================================== 
Reported By:                jtodd
Assigned To:                murf
====================================================================== 
Project:                    Asterisk
Issue ID:                   11233
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89242 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             11-13-2007 11:07 CST
Last Modified:              11-13-2007 12:24 CST
====================================================================== 
Summary:                    SIP channel crashes on new call
Description: 

Attempting a call into a SIP channel (with an invalid extension) causes
Asterisk to crash.  Not sure if this is something that is specific to SIP,
but I have no other channel types to test with at the moment.
====================================================================== 

---------------------------------------------------------------------- 
 jtodd - 11-13-07 12:24  
---------------------------------------------------------------------- 
attaching (in-line, sorry) extensions.conf bloc:

[from-jt]
exten => 1,1,Answer
exten => 1,n(monkeys),Playback(tt-monkeys)
exten => 1,n,NoOp(Channel data pre-hangup: ${CHANNEL(rtpqos,audio,all)})
exten => 1,n,Hangup

exten => h,1,NoOp(Channel data post-hangup: ${CHANNEL(${CHANNEL})})
exten => h,n,Hangup

exten => 222,1,Dial(SIP/65579 at ucla.edu,20)


exten => 111,1,Dial(SIP/s at jt-test.talkplus.com)

exten => 555,1,JabberSend(johnhtodd,testgtalkuser at gmail.com,Invisible man
here sending a message)
exten => 555,n,Dial(Gtalk/johnhtodd/testgtalkuser at gmail.com)
exten => 555,n,Hangup
exten => 666,1,JabberSend(johnhtodd,othertestgmailuser at gmail.com,Invisible
man here sending a message)
exten => 666,n,Hangup

exten => 777,1,Dial(SIP/16502499010 at ms1.dev1)

exten => 888,1,NoOp(Starting ENUM stuff)
exten => 888,n,Set(Foo=${ENUMLOOKUP(43158058304,sip,,1,e164.arpa)})
exten => 888,n,NoOp(Hey, Foo is set to ${Foo})
exten => 888,n,Hangup

exten => 999,1,Set(SIP-USER=testgtalkuser)
exten => 999,n,Goto(gmail.com,s,1)

exten => 123,1,Dial(SIP/jtodd,20)
exten => 123,2,Hangup

exten =>
_1XXXXXXX.,1,Dial(IAX2/3015611020:notmyrealpassword at iax.binfone.com/${EXTEN})


exten => _X*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1) 
exten => _XXXXXXXXXXXXX*.,1,Goto(jt-isn,${EXTEN},1)

exten => _XXXXXXXXXXX,1,Dial(IAX2/3015611020/${EXTEN})) 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-13-07 12:24  jtodd          Note Added: 0073581                          
======================================================================




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