[SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized
Ben Fitzgerald
ben at letscorp.us
Thu Jul 16 19:31:19 CEST 2015
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 at letscorp.us
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On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo <
alberto.sagredo at 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 at 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 at asteriskIP>
>> From: <sip:40081 at asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0
>> CSeq: 10 REGISTER
>> Call-ID: 0005ce130bcee5c4-26538 at kamailio
>> Max-Forwards: 70
>> Content-Length: 0
>> User-Agent: kamailio (4.3.0 (x86_64/linux))
>> Contact: <sip:40081 at 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 at asteriskIP>;tag=32fda68bf54efeeb04e3edc67b53c63d-cfb0
>> To: <sip:40081 at 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 at 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 at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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