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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14841 
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Reported By:                falves11
Assigned To:                
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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:              
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Date Submitted:             2009-04-06 18:32 CDT
Last Modified:              2009-04-07 01:39 CDT
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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

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---------------------------------------------------------------------- 
 (0102812) falves11 (reporter) - 2009-04-07 01:39
 http://bugs.digium.com/view.php?id=14841#c102812 
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I checked another switch, Sansay, and as you can see below, it is perfect.
So Asteris has bug.

INVITE TO PROVIDER:

INVITE sip:823012148761404 at xxx.78.161.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 65.xx.254.xx:5060;branch=z9hG4bK57247301-bdb1079462552
From: <sip:2145551212 at 65.90.254.20>;tag=5724730-fdb1079462552
To: <sip:823012148761404 at xxx.78.161.xx:5060>
Call-ID: 213-1-1239085615 at 65.xx.254.xx
CSeq: 1 INVITE
Server: Sansay VSX 2.1
Contact: <sip:2145551212 at 65.xx.254.xx:5060;transport=udp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length:   257
v=0
o=sansay-VSX 10 10 IN IP4 24.x.44.xxx
s=session controller
c=IN IP4 24.x.44.xxx
t=0 0
m=audio 43456 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-07 01:39 falves11       Note Added: 0102812                          
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