[Asterisk-Users] Asterisk does not handle call from a Cisco IAD correctly

[James Sizemore]

when my Cisco IAD send a call to my Asterisk gateway the gateway treats 
it as if I don't have a peer statement in sip.conf, when I do. Here are 
the first two packets, notice the "Found no matching peer or user for 
'192.168.7.250:50437'" on the second packet. Any one seen this before, 
or have a clue as to the problem?  Asterisk 1.0.9

sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no


Sip read:
INVITE sip:6155555917 at 192.168.53.68:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" <sip:6155552115 at 192.168.7.250>;tag=19D8A640-5E9
To: <sip:6155555917 at 192.168.53.68>
Date: Wed, 06 Mar 2002 00:27:08 GMT
Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8 at 192.168.7.250
Supported: 100rel,timer
Min-SE:  1800
Cisco-Guid: 3128236623-802099670-2154346748-2004044536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: "James Sizemore" 
<sip:6155552115 at 192.168.7.250>;party=calling;screen=yes;privacy=off
Timestamp: 1015374428
Contact: <sip:6155552115 at 192.168.7.250:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 191

v=0
o=CiscoSystemsSIP-GW-UserAgent 6047 8216 IN IP4 192.168.7.250
s=SIP Call
c=IN IP4 192.168.7.250
t=0 0
m=audio 16434 RTP/AVP 0
c=IN IP4 192.168.7.250
a=rtpmap:0 PCMU/8000
a=ptime:20

20 headers, 9 lines
Using latest request as basis request
Sending to 192.168.7.250 : 5060 (non-NAT)
Found no matching peer or user for '192.168.7.250:50437'
Found RTP audio format 0
Peer audio RTP is at port 192.168.7.250:16434
Found description format PCMU
Capabilities: us - 0x78e (gsm|ulaw|alaw|lpc10|g729|speex|ilbc), peer - 
audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined 
- 0x0 (nothing)
Looking for 6155555917 in default
list_route: hop: <sip:6155552115 at 192.168.7.250:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: "James Sizemore" <sip:6155552115 at 192.168.7.250>;tag=19D8A640-5E9
To: <sip:6155555917 at 192.168.53.68>;tag=as43478a8a
Call-ID: BA75B677-2FCF11D6-806AB8FC-77734AF8 at 192.168.7.250
CSeq: 101 INVITE
User-Agent: Memphis ISDN-NET PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:6155555917 at 192.168.53.68>
Content-Length: 0