[SR-Users] Kamailio & Asterisk SIP Registration Forwarding - Asterisk replies 401 Unauthorized

Ben Fitzgerald ben at letscorp.us
Thu Jul 16 23:59:49 CEST 2015 

Thank you for the qualify solution, that worked.

However, on the KB by asipto, they only create a `sipreg` and `sipusers`
table and then in extconfig.conf for asterisk, sipusers and sippeers are
both using the `sipusers` table in MySQL.

I included a sip trace in the original email but I will include a more
detailed sip debug here. It looks like Asterisk and Kamailio can exchange
messages but for some reason, the SIP dialog stops after Asterisk sends
back a SIP 401 Unauthorized to Kamailio. Any ideas?

*1. Kamailio using sipgrep*

T 2015/07/16 14:50:52.393582 UserAgentIP:64521 -> KamailioIP:5060
 [AP]
REGISTER sip:opvpnx.ulets.us SIP/2.0.
Via: SIP/2.0/TCP 192.168.0.179:64521;alias;branch=z9hG4bK.j~V~btADL;rport.
From: <sip:102 at opvpnx.ulets.us>;tag=QZ7de-7u5.
To: sip:102 at opvpnx.ulets.us.
CSeq: 29 REGISTER.
Call-ID: puXkrkIICT.
Max-Forwards: 70.
Supported: outbound.
Accept: application/sdp, text/plain, application/vnd.gsma.rcs-ft-http+xml.
Contact: <sip:102@
 UserAgentIP:64521;transport=tcp>;+sip.instance="<urn:uuid:f8f0aa7c-5b20-4ff2-ac5a-d7b4004afb50>".
Expires: 3600.
User-Agent: Alpha TalkIphone/2.2.5-80-g783bf67 (belle-sip/1.4.0).
Content-Length: 0.
Authorization:  Digest realm="opvpnx.ulets.us",
nonce="VagoaFWoJzylK0MxoOAIPTRhtZBlmVmr", username="102",  uri="sip:
opvpnx.ulets.us", response="24b8f292fca38e72fbcf36417dcecd24".
.


T 2015/07/16 14:50:52.440789 KamailioIP:5060 -> UserAgentIP:64521
 [AP]
SIP/2.0 200 OK.
Via: SIP/2.0/TCP 192.168.0.179:64521
;alias;branch=z9hG4bK.j~V~btADL;rport=64521;received= UserAgentIP.
From: <sip:102 at opvpnx.ulets.us>;tag=QZ7de-7u5.
To: sip:102 at opvpnx.ulets.us;tag=723cfa83f1495d1e63c1f1bb20bde818.a56d.
CSeq: 29 REGISTER.
Call-ID: puXkrkIICT.
Contact: <sip:102@
 UserAgentIP:64521;transport=tcp>;expires=3600;received="sip:
UserAgentIP:64521;transport=tcp";+sip.instance="<urn:uuid:f8f0aa7c-5b20-4ff2-ac5a-d7b4004afb50>".
LETSSBC.
Content-Length: 0.
.

*#*
*# These next two messages when Kamailio forwards REGISTER to Asterisk*
*#*

T 2015/07/16 14:50:52.466461 KamailioIP:43488 -> AsteriskIP:5060
 [AP]
REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0.
Via:
SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0.
To: <sip:102@ AsteriskIP >.
From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497.
CSeq: 10 REGISTER.
Call-ID: 2ee5ec48557bba33-31464@ KamailioIP.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: kamailio (4.3.0 (x86_64/linux)).
Contact: <sip:102@ KamailioIP:5060>.
Expires: 3600.
.


T 2015/07/16 14:50:52.494578 AsteriskIP:5060 -> KamailioIP:43488
 [AP]
SIP/2.0 401 Unauthorized.
Via:
SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0;received=
KamailioIP.
From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497.
To: <sip:102@ AsteriskIP >;tag=as0eb2442e.
Call-ID: 2ee5ec48557bba33-31464@ KamailioIP.
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="5b30f8aa".
Content-Length: 0.

*2. Asterisk using sip set debug on*

t91*CLI>

<--- SIP read from TCP: KamailioIP:43488 --->
REGISTER sip: AsteriskIP:5060;transport=tcp SIP/2.0
Via:
SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0
To: <sip:102@ AsteriskIP >
From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497
CSeq: 10 REGISTER
Call-ID: 2ee5ec48557bba33-31464@ KamailioIP
Max-Forwards: 70
Content-Length: 0
User-Agent: kamailio (4.3.0 (x86_64/linux))
Contact: <sip:102@ KamailioIP:5060>
Expires: 3600

<------------->
--- (11 headers 0 lines) ---
Sending to KamailioIP:5060 (no NAT)
Sending to KamailioIP:5060 (no NAT)

<--- Transmitting (no NAT) to KamailioIP:5060 --->
SIP/2.0 401 Unauthorized
Via:
SIP/2.0/TCP KamailioIP;branch=z9hG4bK328c.29246e24000000000000000000000000.0;received=
KamailioIP
From: <sip:102@ AsteriskIP >;tag=32fda68bf54efeeb04e3edc67b53c63d-3497
To: <sip:102@ AsteriskIP >;tag=as0eb2442e
Call-ID: 2ee5ec48557bba33-31464@ KamailioIP
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="5b30f8aa"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2ee5ec48557bba33-31464@ KamailioIP'
in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog '2ee5ec48557bba33-31464@ KamailioIP'
in 32000 ms (Method: REGISTER)

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

On Thu, Jul 16, 2015 at 11:48 AM, Alberto Sagredo <
alberto.sagredo at avanzada7.com> wrote:

> 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 at 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 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
>>
>> 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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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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