Maybe you got to get some traces with sip set debug on on asterisk or ngrep in kamailio to check whereis the problem.

I think you are not authenticating correctly

Check if you insert on sipusers and sipppers table what is commented on KB by asipto.

Maybe your Kamailio is not responding to OPTIONS (qualify=yes)

add at the beginning of your kamailio.cfg file

request_route {

    if(is_method("OPTIONS") ) {

                sl_send_reply("200","Keepalive");

                exit;   

        }

.....

To solve qualify problem

BR

2015-07-16 19:31 GMT+02:00 Ben Fitzgerald <ben@letscorp.us>:

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@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@avanzada7.com> 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 <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/2.0

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@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

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