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/2.0 Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 To: <sip:40081@asteriskIP> From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 CSeq: 10 REGISTER Call-ID: 0005ce130bcee5c4-26538@kamailio Max-Forwards: 70 Content-Length: 0 User-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 Unauthorized Via: SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= kamailio From: <sip:40081@asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 To: <sip:40081@asteriskIP>;tag=as404bac9a Call-ID: 0005ce130bcee5c4-26538@ kamailio CSeq: 10 REGISTER Server: Asterisk PBX 11.6-cert2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-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=friend dtmfmode=auto host=kamailioIP allow=all context=sipout insecure=port,invite canreinvite=no ======================== Asterisk version: 11.6-cert2 Kamailio version: 4.3 Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Thanks for your response. I did read the section about the secret in the kb url. I followed the example and inserted the test users on tFe url (101, 102, 103) and they have secret set to NULL. I have tried both secret=NULL and secret="" and Asterisk still asks for authentication. Also when I do "sip show peers" I get: Name/username Host Dyn Forcerport ACL Port Status Description Realtime kamailio-inbound kamailioIP a 5060 Unmonitored I added qualify=yes and now: Name/username Host Dyn Forcerport ACL Port Status Description Realtime kamailio-inbound kamailioIP a 5060 UNREACHABLE Could this be the issue? I have verified that Kamailio receives the responses by doing ngrep and I can see the SIP 401 from Asterisk. Maybe I am missing something else? I'm not sure I understand how Asterisk's peer selection affects this. When I received the registration request from Kamailio, the From: address and domain are the same as the To: address and domain, which are the values I have set in the sipusers table. Another thing, even though the client handset says registered, the table 'sipregs' is not updated with fullcontact, regseconds, or any data at all. Yet I can still make a call. So maybe Asterisk is not authenticating INVITES (whether or not it's registered) and that's why I can call. Any further help or things I should try? Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600 On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo < alberto.sagredo(a)avanzada7.com&gt; wrote: > You could remove secret= on extensiones to check if its related to > authentication or not > > You must not request authentication to kamailio in order to work > properly in front of Asterisk > > As Daniel mention check if Kamailio peer is created and extensiones > have no secret.. you would need to add alternate sippasswd table for > kamailio authentication > > BR > > 2015-07-16 1:42 GMT+02:00 Ben Fitzgerald &lt;ben(a)letscorp.us&gt;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/2.0 >> Via: SIP/2.0/TCP >> kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0 >> To: <sip:40081@asteriskIP> >> From: <sip:40081@asteriskIP >> >;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >> CSeq: 10 REGISTER >> Call-ID: 0005ce130bcee5c4-26538@kamailio >> Max-Forwards: 70 >> Content-Length: 0 >> User-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 Unauthorized >> Via: >> SIP/2.0/TCP kamailio;branch=z9hG4bK998f.2846e405000000000000000000000000.0;received= >> kamailio >> From: <sip:40081@asteriskIP >> >;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0 >> To: <sip:40081@asteriskIP>;tag=as404bac9a >> Call-ID: 0005ce130bcee5c4-26538@ kamailio >> CSeq: 10 REGISTER >> Server: Asterisk PBX 11.6-cert2 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH >> Supported: replaces, timer >> WWW-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=friend >> dtmfmode=auto >> host=kamailioIP >> allow=all >> context=sipout >> insecure=port,invite >> canreinvite=no >> >> ======================== >> >> Asterisk version: 11.6-cert2 >> Kamailio version: 4.3 >> >> Benjamin Fitzgerald >> LETS Corporation >> (925) 235-1154 >> ben(a)letscorp.us >> >> >> >> >> *******Confidential Notice: >> This message is intended only for the use of the individual or entity >> to which it is addressed and may contain information that is privileged, >> confidential and exempt from disclosure under applicable law. If the reader >> of this message is not the intended recipient, you are hereby notified that >> any dissemination, distribution or copying of this communication is >> strictly prohibited. If you have received this message in error, please >> delete this message from all computers and contact Orion Systems/LETS Corp >> immediately by return e-mail and/or telephone at (925) 566-5600 >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users