16 Jul
2015
16 Jul
'15
9:48 p.m.
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
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On Thu, Jul 16, 2015 at 3:40 AM, Alberto Sagredo <
alberto.sagredo(a)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(a)letscorp.us>us>:
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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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