Administrator TOOTAI
2006-Jan-27 01:57 UTC
[Asterisk-Users] No matching peer or user based on IP address
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@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