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 - [EMAIL PROTECTED]
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
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