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

[Asterisk Bug Tracker]

The following issue has been RESOLVED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16061 
====================================================================== 
Reported By:                dveiga
Assigned To:                russell
====================================================================== 
Project:                    Asterisk
Issue ID:                   16061
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
Asterisk Version:           SVN 
JIRA:                       SWP-921 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 223801 
Request Review:              
Resolution:                 won't fix
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-10-12 18:08 CDT
Last Modified:              2010-03-15 10:32 CDT
====================================================================== 
Summary:                    [patch] Peer mismatch in incomming call
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).
====================================================================== 

---------------------------------------------------------------------- 
 (0119359) russell (administrator) - 2010-03-15 10:32
 https://issues.asterisk.org/view.php?id=16061#c119359 
---------------------------------------------------------------------- 
Thank you for your contribution.  However, this change can not be accepted.
 This is a problem that people have solved via different configuration
methods.  There has also been much vigorous architectural discussion on the
best way to improve matching in chan_sip.  Regardless of what is done, it
can _not_ be done as changes in behavior to the existing matching behavior
of type=user and type=peer.    If you're interested in joining the
discussion on the design of future matching methods, I would encourage you
to join the asterisk-dev list, and look up previous discussions that have
occurred in the list archives. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-15 10:32 russell        Note Added: 0119359                          
2010-03-15 10:32 russell        Status                   confirmed => resolved
2010-03-15 10:32 russell        Resolution               open => won't fix   
2010-03-15 10:32 russell        Assigned To               => russell         
======================================================================