[asterisk-bugs] [Asterisk 0016061]: [patch] Peer mismatch in incomming call [PATCH]

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16061 
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Reported By:                dveiga
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16061
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 223801 
Request Review:              
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Date Submitted:             2009-10-12 18:08 CDT
Last Modified:              2009-10-14 12:22 CDT
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Summary:                    [patch] Peer mismatch in incomming call [PATCH]
Description: 
    I'm using the latest 1.4 svn version as a voip gateway connected to a
PBX. As such I have as many DAHDI channels as SIP peers and link them on a
one to one basis. When commertial gateways (say Grandstream 4024) register
to a PBX, they use different local SIP ports for each channel. In this way,
when the PBX places a SIP invite to offer a call using a certain port, ther
is no doubt which SIP peer the call is for.
    Asterisk cannot use different ports, so upon receiving an invite, it
has to determine which peer it is for. The default behaviour is to
determine the peer based on the IP address (after trying to find a user,
that is not applicable here), but when there are multiple subscriptions to
the same IP this is not enough.
    The current patch analyzes the sip headers to determine the peer name
and then finds the peer based on both: IP an NAME. For compatibility
reasons, if a peer matching both is not found, it falls back to the
previous method (finding one just using the IP address).
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 (0112273) ebroad (manager) - 2009-10-14 12:22
 https://issues.asterisk.org/view.php?id=16061#c112273 
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Correct me if I am wrong:

;sip.conf
[100]
host=foo
callerid=100

[101]
host=foo
callerid=101

[102]
host=bar

If 102 places a call to 101 or 100, does the call complete? is it coming
in on the right line? 

Note: I had a (possibly) similar issue with some Cisco 7940's on a local
network segment, where line 1 would use port 5060 on the phone and line 2
would use some random high port. This would cause registrations for line 2
to fail because Asterisk would challenge or ACK the registration request on
port 5060. Enabling NAT support on the phone forced it to use 5060 for both
lines, resolving the issue.
 

Issue History 
Date Modified    Username       Field                    Change               
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2009-10-14 12:22 ebroad         Note Added: 0112273                          
2009-10-14 12:22 ebroad         Status                   new => feedback     
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