17 Jul
2015
17 Jul
'15
8:38 p.m.
I think I have fixed the authentication issue yet the SIP dialog has
completely changed. Now the dialog involves Asterisk sending SIP NOTIFY to
Kamailio, which is then forwarded to the client. Kamailio.cfg has no routes
to handle NOTIFY and there are no SUBSCRIBE messages preceding the NOTIFY.
Only REGISTER and 200 OK. Is this expected behavior? The sipregs database
is now correctly updated when a peer registers so that's good.
Benjamin Fitzgerald
LETS Corporation
(925) 235-1154
ben(a)letscorp.us
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On Thu, Jul 16, 2015 at 2:59 PM, Ben Fitzgerald <ben(a)letscorp.us> wrote:
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@opvpnx.ulets.us>;tag=QZ7de-7u5.
To: sip:102@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@opvpnx.ulets.us>;tag=QZ7de-7u5.
To: sip:102@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(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 11:48 AM, Alberto Sagredo <
alberto.sagredo(a)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(a)letscorp.us>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(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> wrote:
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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(a)letscorp.us>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
>
>
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_______________________________________________
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