[asterisk-bugs] [Asterisk 0018338]: [patch] (Call Completion / SIP) INVITE Fails (Receive a 404 From Asterisk Server) When Using The URI Provided From A NOTIFY(cc-r

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18338 
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Reported By:                GeorgeKonopacki
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   18338
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
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:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-11-19 08:40 CST
Last Modified:              2010-12-20 15:40 CST
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Summary:                    [patch] (Call Completion / SIP) INVITE Fails
(Receive a 404 From Asterisk Server) When Using The URI Provided From A
NOTIFY(cc-r
Description: 
The ‘To’ line in our INVITE contains:

To:
<sip:10f2a43020db0a2c787fa67c6c4d279b at 192.168.233.250:5060;transport=udp>

The function in ‘find_by_notify_uri_helper’ in file chan_sip.c does a
straight strcmp of the ‘To’ with the URI it sent in the NOTIFY
(cc-ready).

This match then fails because we have added ';transport=udp'

return !strcmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;

agent_pvt->notify_uri =
sip:10f2a43020db0a2c787fa67c6c4d279b at 192.168.233.250:5060

uri =
sip:10f2a43020db0a2c787fa67c6c4d279b at 192.168.233.250:5060;transport=udp

The Asterisk server is NOT extracting the URI correctly from the ‘To’
line in the INVITE. So strcmp will always fail.
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---------------------------------------------------------------------- 
 (0129813) svnbot (reporter) - 2010-12-20 15:40
 https://issues.asterisk.org/view.php?id=18338#c129813 
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Repository: asterisk
Revision: 299249

_U  trunk/
U   trunk/channels/chan_sip.c

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r299249 | mmichelson | 2010-12-20 15:40:33 -0600 (Mon, 20 Dec 2010) | 25
lines

Merged revisions 299248 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec 2010) | 20
lines
  
  Fix a couple of CCSS issues.
  
  * Make sure to allocate a cc_params structure
    when creating autopeers.
  
  * Use sip_uri_cmp when retrieving SIP CC agents
    and monitors in case parameters appear in the
    URI.
  
  (closes issue https://issues.asterisk.org/view.php?id=18504)
  Reported by: kkm
  
  (closes issue https://issues.asterisk.org/view.php?id=18338)
  Reported by: GeorgeKonopacki
  Patches: 
        18338.diff uploaded by mmichelson (license 60)
  Tested by: GeorgeKonopacki
........

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http://svn.digium.com/view/asterisk?view=rev&revision=299249 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-20 15:40 svnbot         Checkin                                      
2010-12-20 15:40 svnbot         Note Added: 0129813                          
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