[asterisk-bugs] [Asterisk 0012761]: chan_sip: build_contact() does not put alternate port setting in Contact header
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri May 30 15:38:06 CDT 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=12761
======================================================================
Reported By: asbestoshead
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 12761
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 118255
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 05-30-2008 03:04 CDT
Last Modified: 05-30-2008 15:38 CDT
======================================================================
Summary: chan_sip: build_contact() does not put alternate
port setting in Contact header
Description:
build_contact() looks at the *other* end's port to decide whether there's a
non-standard port in use. If my end has bindport=5062 but the peer has
port=5060, when I send an invite to the peer, my Contact header doesn't
have ":5062".
initreqprep() has the same problem when setting the From: header.
chan_sip in Asterisk trunk and 1.6.0-beta8 (but not 1.4.19) have this
problem.
I have a trivial patch for this against Moy's mfcr2 branch off trunk, but
I can't find the link to sign the contributor's agreement. The patch also
applies to 1.6.0beta8, although I haven't tested it there...
======================================================================
----------------------------------------------------------------------
rjain - 05-30-08 15:38
----------------------------------------------------------------------
Yes, that was it. I see the problem now after I added SIPp to sip.conf as a
peer (instead of directly dialing to it by IP). I'm not sure how this broke
from 1.4 to 1.6. The way you're proposing fixing it, looks good, but we'll
have to think if that can impact any other scenarios. Ideally, it would be
nice if we can backtrace this to when the bug was introduced. Are you still
planning on uploading your patch?
Issue History
Date Modified Username Field Change
======================================================================
05-30-08 15:38 rjain Note Added: 0087577
======================================================================
More information about the asterisk-bugs
mailing list