[asterisk-users] Could Asterisk be rewriting an incoming invite?

[Chris Garrigues]

I've just determined that it IS happening on my box, but why?
I did a packet capture using tcpdump on this very same box and it shows the
correct invite while sip debug shows the wrong values.  here's what I see in
wireshark:

No.     Time        Source                Destination           Protocol
Info
      1 0.000000    216.82.224.202        67.198.16.18          SIP/SDP
Request: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp, with
session description

Frame 1 (1043 bytes on wire, 1043 bytes captured)
Ethernet II, Src: EciTelec_00:a0:41 (00:02:0e:00:a0:41), Dst: Intel_92:3b:be
(00:0c:f1:92:3b:be)
Internet Protocol, Src: 216.82.224.202 (216.82.224.202), Dst: 67.198.16.18
(67.198.16.18)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp
SIP/2.0
    Message Header
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1237225281 1237225282 IN IP4
209.244.187.171
            Session Name (s): -
            Connection Information (c): IN IP4 209.244.187.171
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 60570 RTP/AVP 0
18 101
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15

and here's what I see in sip debug:

INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0
Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>
Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>
Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK525.4ab0348.0
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK525.3b6e7ab3.0
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207517079314
From: "GARRIGUES,CHRIS"
<sip:+15124990483 at 4.68.250.148<sip%3A%2B15124990483 at 4.68.250.148>
;isup-oli=0>;tag=VPSF506071629460
To: <sip:+15129616808 at 4.79.212.229:5060>
Call-ID: HOUMGC0520090316174121064302 at 209.244.63.35
CSeq: 1 INVITE
Contact: <sip:+15124990483 at 216.82.224.202:5060;transport=udp>
Max-Forwards: 67
Content-Type: application/sdp
Content-Length: 175
Remote-Party-ID: "GARRIGUES,CHRIS"
<sip:+15124990483 at 4.68.250.148<sip%3A%2B15124990483 at 4.68.250.148>
>;party=calling;screen=yes;privacy=off

v=0
o=- 1237225281 1237225282 IN IP4 216.82.224.202
s=-
c=IN IP4 216.82.224.202
t=0 0
m=audio 60570 RTP/AVP 0 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15


What is rewriting my o= and c=


??????
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