You could remove secret= on extensiones to check if its related to authentication or notYou must not request authentication to kamailio in order to work properly in front of AsteriskAs Daniel mention check if Kamailio peer is created and extensiones have no secret.. you would need to add alternate sippasswd table for kamailio authentication
BR2015-07-16 1:42 GMT+02:00 Ben Fitzgerald <ben@letscorp.us>:_______________________________________________Hi, I've been following this integration tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and have a successful registration and I can even make calls through my asterisk box.However what is unusual to me is that every time a phone registers with Kamailio, that is forwarded to Asterisk (as expected), yet Asterisk replies with 401 Unauthorized. Oddly enough the phone registers and can still make calls. What worries me is that as we scale to 100's of cps, this seemingly erroneous message may slow down Asterisk because it's trying to handle authentication for users which have already been authenticated by Kamailio. If this behavior is expected, then that would be good to know as well.This is the sip debug from ASTERISK (I have replaced IP's with the names of the servers):<--- SIP read from TCP:kamailio:41205 --->REGISTER sip:asteriskIP:5060;transport=tcp SIP/2.0Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0To: <sip:40081@asteriskIP>From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0CSeq: 10 REGISTERCall-ID: 0005ce130bcee5c4-26538@kamailioMax-Forwards: 70Content-Length: 0User-Agent: kamailio (4.3.0 (x86_64/linux))Contact: <sip:40081@kamailio:5060>Expires: 3600<------------->--- (11 headers 0 lines) ---Sending to kamailio:5060 (no NAT)Sending to kamailio:5060 (no NAT)<--- Transmitting (no NAT) to kamailio:5060 --->SIP/2.0 401 UnauthorizedVia: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailioFrom: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0To: <sip:40081@asteriskIP>;tag=as404bac9aCall-ID: 0005ce130bcee5c4-26538@ kamailioCSeq: 10 REGISTERServer: Asterisk PBX 11.6-cert2Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="262b338e"Content-Length: 0<------------>Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER)Scheduling destruction of SIP dialog '0005ce130bcee5c4-26538@ kamailio' in 32000 ms (Method: REGISTER)Really destroying SIP dialog '0005ce130bcee5c1-26536@ kamailio' Method: REGISTER=========================sip.conf for kamailio trunk:[kamailio-inbound]type=frienddtmfmode=autohost=kamailioIPallow=allcontext=sipoutinsecure=port,invitecanreinvite=no========================Asterisk version: 11.6-cert2Kamailio version: 4.3Benjamin FitzgeraldLETS Corporation
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