asterisk users - Nov 2004 - Asterisk using the wrong peer in sip.conf

I'm having a problem with Asterisk choosing the wrong peer entry from the
sip.conf file.  Based off the debug below Asterisk should see the message
coming from TNT3 (see the sip.conf below) and not from SER_FAX (which it
shows in the debug below "Found peer 'SER_FAX'").  Based off
what I have
here it seems to me that since the From is 198.88.216.30 it should match
with the TNT3 entry in my sip.conf which is what I want it to do, instead it
is matching with the SIP proxy that is proxying it the SIP message.  Is
there a way to get Asterisk to lookup based of the originator of the INVITE
instead of by who last proxied it the INVITE?

Basically here is my setup:

198.88.216.84 = SER
198.88.216.30 = TNT3 (My PSTN Gateway)
198.88.216.85 = Asterisk

TNT3 <---> SER Proxy <---> Asterisk




----- Here is the debug from Asterisk:

Sip read: 
INVITE sip:4444444444@ast.bwsys.net:5060;user=phone SIP/2.0
Record-Route:
<sip:4444444444@198.88.216.84:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on>
To:   <sip:4444444444@198.88.216.84:5060;user=phone>
From:
<sip:5555555555@198.88.216.30:5060;user=phone>;tag=46cae657-1bf8d0c3-1ed858c
6
Remote-Party-Id:
<sip:5555555555@198.88.216.30:5060;user=phone>;screen=yes;id-type=subscriber
;party=calling;privacy=off
Proxy-Require: privacy
Call-ID: 83221e88-236af-1bf8d0c3@198.88.216.30
CSeq: 93145626 INVITE
Via: SIP/2.0/UDP 198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0
Via: SIP/2.0/UDP 198.88.216.30:5060;rport=5060
Max-Forwards: 69
Contact: <sip:5555555555@198.88.216.30:5060;user=phone>
Supported: replaces
Supported: 100rel
Content-Type: application/sdp
Content-Length: 272

v=0
o=TNT3 469291203 469291203 IN IP4 198.88.216.30
s=Session SDP
c=IN IP4 198.88.216.30
t=0 0
m=audio 44518 RTP/AVP 18 0 8 96 
a=silenceSupp:off
a=ecan:b on g168
a=rtpmap:96 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000

16 headers, 12 lines
Using latest request as basis request
Sending to 198.88.216.84 : 5060 (non-NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Peer audio RTP is at port 198.88.216.30:44518
Found description format telephone-event
Found description format PCMA
Found description format PCMU
Found description format G729
Capabilities: us - 0x10c(ULAW|ALAW|G729A), peer -
audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined -
0x10c(ULAW|ALAW|G729A)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found peer 'SER_FAX'
Looking for 4444444444 in FROM_PSTN
list_route: hop:
<sip:4444444444@198.88.216.84:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on>
list_route: hop: <sip:5555555555@198.88.216.30:5060;user=phone>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0;received=198.88.216.84;rport=506
0
Via: SIP/2.0/UDP 198.88.216.30:5060
From:
<sip:5555555555@198.88.216.30:5060;user=phone>;tag=46cae657-1bf8d0c3-1ed858c
6
To: <sip:4444444444@198.88.216.84:5060;user=phone>;tag=as62f171b3
Call-ID: 83221e88-236af-1bf8d0c3@198.88.216.30
CSeq: 93145626 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:4444444444@198.88.216.85>
Content-Length: 0






----- Here is my sip.conf:

[general]
port=5060                       ; Port to bind to
bindaddr=0.0.0.0                ; Address to bind to
context=FROM_PSTN               ; Default for incoming calls
rtptimeout=30                   ; Terminate call if 60 seconds of no RTP
activity
canreinvite=no
disallow=all
allow=g729
allow=ulaw
allow=alaw

[SER]
type=friend
host=198.88.216.84
port=5060
qualify=no
nat=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw


[SER_FAX]
type=friend
host=198.88.216.84
port=5060
qualify=no
nat=yes
disallow=all
allow=ulaw


[TNT3]
type=friend
host=198.88.216.30
port=5060
dtmfmode=rfc2833
qualify=no
canreinvite=no
nat=no
context=FROM_PSTN
deny=0.0.0.0
permit=198.88.216.30/255.255.255.255
disallow=all
allow=g729
allow=ulaw
allow=alaw



----------------------------------------

Michael Shuler

First, please do not use "friends". It will confuse you. We receive
calls from type=user
and send calls to type=peer (that is the general idea, anyway... :-)

For an incoming call, we first match on users based on the username part of the
From:
sip address. So if the call comes from sip:oej@edvina.net, we'll match on
"oej".
If we can't find a matching user, we will match on peers based on IP
address, which
is your problem.

Make sure you have a type=user with the correct user name and you should be all
set.

/O