[Asterisk-Users] SIP URI modified unexpectedly! Is that a router problem?

[Dan Zhou]

Hi,

I set up an asterisk server at my home computer, and the * box is configured 
as the DMZ of my ADSL modem/router. I found the SIP URI in an INVITE message 
has been changed before it reaches the * server.
In my setup,  I have a SJPhone software installed in with a public IP 
yy.yy.yy.yy.  (ph no. 10930)
I have a unknown brand SIP phone in the same network as the * sever is in. 
The public IP is xx.xx.xx.xx, phone no =10916.


Here is the SJphone Log of an INVITE message sent from 10930 to 10916.
10:42:05 INFO Initiating SIP call to sip:10916 at xx.xx.xx.xx:5060
10:42:05 DEBUG
2005-02-05 21:42:05.937 UDP LOCAL->xx.xx.xx.xx:5060
INVITE sip:10916 at xx.xx.xx.xx:5060 SIP/2.0
Content-Length: 341
Contact: <sip:10930 at yy.yy.yy.yy:5060>
Call-ID: C64037DF-A3BA-4F1D-8FE9-84FA9D3DBD85 at yy.yy.yy.yy
Content-Type: application/sdp
From: "Dan"<sip:10930 at xx.xx.xx.xx:5060>;tag=588746825471
CSeq: 1 INVITE
Max-Forwards: 70
To: <sip:10916 at xx.xx.xx.xx:5060>
Via: SIP/2.0/UDP 
yy.yy.yy.yy;rport;branch=z9hG4bKcb617ac20131c9b142053dad0000798300000032
User-Agent: SJLabs-SJphone/1.40.258
--snip---


Here is the corresponding message I received in my server, the output of 
"ngrep 10930 port 5060 -d eth0" .

###
U yy.yy.yy.yy:5060 -> 192.168.1.2:5060
INVITE sip:192.168.1.2 at xx.xx.xx.xx:5060 SIP/2.0..Content-Length: 341..Co
ntact: <sip:10930 at yy.yy.yy.yy:5060>..Call-ID: C64037DF-A3BA-4F1D-8FE9-84
FA9D3DBD85 at yy.yy.yy.yy..Content-Type: application/sdp..From: "Dan"<sip:1
0930 at xx.xx.xx.xx:5060>;tag=588746825471..CSeq: 1 INVITE..Max-Forwards: 7
0..To: <sip:192.168.1.2 at xx.xx.xx.xx:5060>..Via: SIP/2.0/UDP 203.97.122.1
94;rport;branch=z9hG4bKcb617ac20131c9b142053dad0000798300000032..User-Agent
: SJLabs-SJphone/1.40.258....v=0..o=- 3316628525 3316628525 IN IP4 203.97.1
22.194..s=SJphone..c=IN IP4 yy.yy.yy.yy..t=0 0..a=direction:active..m=au
dio 16386 RTP/AVP 0 8 3 97 98 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/80
00..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:98 iLBC/8000..a=fm
tp:98 mode=20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16..
#

Obviously, they are two places in the received messages changed (the INVITE 
tag and the "To" header), from
<sip:10916 at xx.xx.xx.xx:5060>
to
sip:192.168.1.2 at xx.xx.xx.xx:5060

As a result, I got a 404 not found error. Before the error, there is one 
line in the CLI output saying:
Looking for 192.169.1.2 in local ...
I think it should have looked for 10916 in local (the context defined in 
extension.conf).
Has anyone here experienced similar problem? Can I say my router is not VoIP 
friendly?

Cheers,
Dan

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