[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 13:27:01 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 13:27 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...
======================================================================
----------------------------------------------------------------------
asbestoshead - 05-30-08 13:27
----------------------------------------------------------------------
Hi rjain, I think there are some differences with our setups:
- both your local and remote SIP endpoints are on non-standard ports;
- your clients look like they're behind NAT;
- your clients are registering to your Asterisk, which may trigger a code
path where p->socket.port gets filled in correctly with Asterisk's bind
port.
Mike
Issue History
Date Modified Username Field Change
======================================================================
05-30-08 13:27 asbestoshead Note Added: 0087564
======================================================================
More information about the asterisk-bugs
mailing list