[asterisk-bugs] [Asterisk 0018336]: (Call Completion / SIP) No response is received if we try to subscribe for call completion after we have received a 180 Ringing

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Dec 6 09:26:23 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18336 
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Reported By:                GeorgeKonopacki
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18336
Category:                   Channels/chan_sip/Subscriptions
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-11-19 08:20 CST
Last Modified:              2010-12-06 09:26 CST
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Summary:                    (Call Completion / SIP) No response is received if
we try to subscribe for call completion after we have received a 180 Ringing
Description: 
The server does NOT send a response if we try to subscribe for call
completion after we have received a 180 Ringing.
You can only subscribe for call completion when the call has been
cleared.

This appears to be a bug is the Asterisk call completion state machine.
When we receive the 180 Ringing, for this call, its call-completion state
is ‘CC_AVAILABLE’. If we then send a subscribe message to the server,
it trys to change the call-completion state to ‘CC_CALLER_REQUESRED’.
Because this is an invalid state change, it just ignores the message. The
only state the Asterisk server will accept our subscribe message is in the
state ‘CC_CALLER_AVAILABLE’.

The Asterisk server will go into the ‘CC_CALLER_OFFERED’ when the SIP
client clears the call by sending a CANCEL.
====================================================================== 

---------------------------------------------------------------------- 
 (0129328) mmichelson (administrator) - 2010-12-06 09:26
 https://issues.asterisk.org/view.php?id=18336#c129328 
---------------------------------------------------------------------- 
There is a new SVN branch located at

http://svn.digium.com/svn/asterisk/team/gruop/ccss_failure_response

that should address both this issue and
https://issues.asterisk.org/view.php?id=18337. The reason it's in an SVN
branch is that it required SIP changes from me as well as ISDN changes from
rmudgett. I've made the necessary SIP changes, but I don't believe the ISDN
changes are there yet. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-06 09:26 mmichelson     Note Added: 0129328                          
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