[asterisk-bugs] [Asterisk 0018917]: Asterisk ignores reInvite

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18917 
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Reported By:                aleksander2011
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18917
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.17 
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:             2011-03-03 08:56 CST
Last Modified:              2011-03-04 01:58 CST
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Summary:                    Asterisk ignores reInvite
Description: 
I do call from host1 (82.204.187.9) to Asterisk (192.168.0.28 - without
NAT), and Asterisk sends this call to host2 (82.204.187.10). When customer
is answering the call, then Asterisk connects RTP directly from host1 to
host2. It work perfectly. 
But If hostB wants send a reInvite to Asterisk (for example put on hold)
and changes  media port, then the reinvite does't reach to host1 (lost in
Asterisk). Asterisk received reInvite don't sent they in other leg. Why?
Option "canreinvite" set "yes" for all peer.
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 (0132628) aleksander2011 (reporter) - 2011-03-04 01:58
 https://issues.asterisk.org/view.php?id=18917#c132628 
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Hello lmadsen.

I have the following topology.
82.204.187.9 (Softswitch Essentra CX 4 class)
82.204.187.7 (Media Gateway)
82.204.187.10 (Softswitch Essentra BAX 5 class)
192.168.0.28 - Asterisk

82.204.187.9 -SIP-> 
                     192.168.0.28 -SIP+RTP->82.204.187.10 -SIP+RTP-> End
User
82.204.187.7 -RTP->

I call to number +74956986015. End user +74956986015 is behind a
82.204.187.10. If he puts on hold incoming call or forwards call to another
user then 82.204.187.10 changes RTP port and sends reinvite.
End user is behind a 82.204.187.10

I attached SIP trace from Asterisk console.
Asterisk received reinvites from 82.204.187.10 in line 1715 and 1888 (put
on hold and forward), but Asterisk didn't send they to 82.204.187.9. And
because of this after forward one way audio. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-04 01:58 aleksander2011 Note Added: 0132628                          
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