[asterisk-bugs] [Asterisk 0014230]: [patch] Calls are not accepted from an outbound proxy

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14230 
====================================================================== 
Reported By:                Nick_Lewis
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   14230
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-01-13 10:19 CST
Last Modified:              2009-01-15 07:11 CST
====================================================================== 
Summary:                    [patch] Calls are not accepted from an outbound
proxy
Description: 
find_peer does not make a match between a received invite and a peer if the
invite comes from the outboundproxy rather than directly from the host. 

For example if a peer has the following sip.conf:

[mytrunk]
host=sip.myitsp.net
port=5060
outboundproxy=nat.myitsp.net:5082

an invite from nat.myitsp.net:5082 is not matched to peer mytrunk

This is a problem for applications in which a natproxy is used for both
incoming and outgoing calls. 
====================================================================== 

---------------------------------------------------------------------- 
 (0097877) Nick_Lewis (reporter) - 2009-01-15 07:11
 http://bugs.digium.com/view.php?id=14230#c97877 
---------------------------------------------------------------------- 
What needs to be separated in the dialplan? I assume we are talking here
about incoming calls only. In the example there is only one voip line from
each ITSP and therefore only one voip line with a particular natproxy. Does
the need for separation in the dialplan apply only to when there are
multiple voip lines sharing the same ITSP and therefore sharing the same
natproxy? 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-15 07:11 Nick_Lewis     Note Added: 0097877                          
======================================================================