[asterisk-bugs] [Asterisk 0015800]: [patch] Fetching SIP headers from BYE sent by callee

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15800 
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Reported By:                sergee
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15800
Category:                   Channels/chan_sip/NewFeature
Reproducibility:            have not tried
Severity:                   feature
Priority:                   normal
Status:                     confirmed
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 214898 
Request Review:              
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Date Submitted:             2009-08-31 08:04 CDT
Last Modified:              2010-03-18 12:30 CDT
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Summary:                    [patch] Fetching SIP headers from BYE sent by callee
Description: 
We have simple situation:

exten => _X.,1,DIAL(SIP/mypeer/${EXTEN})
exten => h,1,AGI(billing)

There is no way to get headers from BYE message sent by callee. But a lot
of harware provide usefull headers in BYE message, for example Linksys
provide header P-RTP-Stat which contain packet counters, jitter, delay
etc..:

P-RTP-Stat:
PS=256,OS=40960,PR=244,OR=39040,PL=29920,JI=5,LA=0,DU=4,EN=G711a,DE=G711a

Cisco provides another usefull header 'Reason' which contain ISDN hangup
cause code:

Reason: Q.850;cause=16


This patch allows user to fetch any header from BYE sent by callee. It
introduces special variable SIPREMOTEBYEHEADER. This variable should
contain 2 coma separated values. 1st value - a sip header that should be
fetched, 2nd value - name of variable to put sip header in.

So to fetch Header 'Reason' sent by cisco, you simply need to set
variable:

exten => _X.,1,SET(SIPREMOTEBYEHEADER=Reason,MYVAR)
exten => _X.,2,DIAL(SIP/mypeer/${EXTEN})
exten => h,1,AGI(billing)

Right after mypeer sends a BYE, you will have variable MYVAR defined on
your channel.

Patches against branch 1.6.0 and trunk provided.
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 (0119557) coolmig (reporter) - 2010-03-18 12:30
 https://issues.asterisk.org/view.php?id=15800#c119557 
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Any updates on this? This could be very helpful to gather the IDSN hangup
cause sent from SIP providers, as the original poster said on the
description. Or there is another method to do this that I missed? 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-18 12:30 coolmig        Note Added: 0119557                          
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