[asterisk-bugs] [Asterisk 0014841]: Outbound Invite has the from field and Remote-Party-ID bad

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Apr 6 20:11:45 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14841 
====================================================================== 
Reported By:                falves11
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14841
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 186654 
Request Review:              
====================================================================== 
Date Submitted:             2009-04-06 18:32 CDT
Last Modified:              2009-04-06 20:11 CDT
====================================================================== 
Summary:                    Outbound Invite has the from field and 
Remote-Party-ID bad
Description: 
I found something that affects interoperability badly. I am sending an
invite to IP 192.168.1.221, from an Asterismk with IP 10.1.1.10. The From
header and the Remote-Party-ID headers have the wrong IP, it should be the
Asterisk, the sender's IP, not the target IP. I checked in the RFC and it
is clear that the from header contains the originator's IP. Several of my
carriers stopped working. I need to keep using this version, because it
does not blow up, SVN-branch-1.6.2-r186654M , instead of any other one.
Somebody please fix it.

INVITE sip:823013212069848 at 192.168.1.221 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.10:5060;branch=z9hG4bK2ebb3639;rport
Max-Forwards: 70
From: "3614296471" <sip:3614296471 at 192.168.1.221>;tag=as02da3780 Should be
from our IP.
To: <sip:823013212069848 at 192.168.1.221>
Contact: <sip:3614296471 at 10.1.1.10>
Call-ID: 4f7b5aa40310f3871f3995de46a1137c at 192.168.1.221
CSeq: 102 INVITE
User-Agent: Cisco 3845
Remote-Party-ID: "3614296471"
<sip:3614296471 at 192.168.1.221>;privacy=off;screen=yes Should be from our
IP.
Date: Mon, 06 Apr 2009 21:51:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
P-Asserted-Identity: "" <sip:+13614296471 at 10.1.1.10>
Content-Type: application/sdp
Content-Length: 295

====================================================================== 

---------------------------------------------------------------------- 
 (0102807) falves11 (reporter) - 2009-04-06 20:11
 http://bugs.digium.com/view.php?id=14841#c102807 
---------------------------------------------------------------------- 
I checked version 1.4 and in fact it has the same issue. I have a vendor,
Verizon, which is maybe the most important US carrier, who requires this
issue fixed. If it breakes anything else then we should make it into
configurable option, pero peer. But the RFC is clear about this. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-06 20:11 falves11       Note Added: 0102807                          
======================================================================




More information about the asterisk-bugs mailing list