16 Jul
2015
16 Jul
'15
1:40 p.m.
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
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