[Asterisk-Users] No matching peer or user based on IP address

[Administrator TOOTAI]

Hi all,

I'm running Asterisk SVN-trunk-r8643M and face following problem:

I'm trying to get incoming call from a provider and calls ended with a 
404 error. On the INVITE I get "Found no matching peer or user for <IP 
address>:5060" and then "Looking for <UserName> in <SIP default context> 
(domain xxx.xxx.xxx.xxx)". My question is why asterisk doesn't found my 
peer/user chapter?

If I add an extension <UserName>,1,blablabla in my SIP default context, 
it's working. The provider has multiple IP address. Here is sip.conf and 
debug logs:

[UserName]
type=user				;tested with friend
username=UserName
fromuser=UserName
fromdomain=ProviderDomain
secret=MySecret
context=from-provider
host=sip.ProviderDomain.com
insecure=port,invite			;tested with very
nat=no
canreinvite=no
disallow=all
allow=alaw,ulaw ;g726

Jan 27 00:42:44 VERBOSE[16980] logger.c: --- (11 headers 8 lines)Jan 27 
00:42:44 VERBOSE[16980] logger.c: --- (11 headers 8 lines)---
Jan 27 00:42:44 VERBOSE[16980] logger.c: Using INVITE request as basis 
request - fd26b95042a345c594bc469b8b4ff9f4 at xxx.xxx.xxx.xxx
Jan 27 00:42:44 VERBOSE[16980] logger.c: Sending to xxx.xxx.xxx.xxx : 
5060 (non-NAT)
Jan 27 00:42:44 VERBOSE[16980] logger.c: Found no matching peer or user 
for 'xxx.xxx.xxx.xxx:5060'
Jan 27 00:42:44 VERBOSE[16980] logger.c: Found RTP audio format 8
Jan 27 00:42:44 VERBOSE[16980] logger.c: Peer audio RTP is at port 
yyy.yyy.yyy.yyy:11274
Jan 27 00:42:44 VERBOSE[16980] logger.c: Found description format pcma
Jan 27 00:42:44 VERBOSE[16980] logger.c: Capabilities: us - 0x40e 
(gsm|ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), 
combined - 0x8 (alaw)
Jan 27 00:42:44 VERBOSE[16980] logger.c: Non-codec capabilities: us - 
0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Jan 27 00:42:44 VERBOSE[16980] logger.c: Looking for <UserName> in <SIP 
default context> (domain xxx.xxx.xxx.xxx)
Jan 27 00:42:44 VERBOSE[16980] logger.c: Reliably Transmitting (no NAT) 
to xxx.xxx.xxx.xxx:5060:
SIP/2.0 404 Not Found

Thank's for any hint
-- 
Daniel