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

[Alberto Sagredo]

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
>
>
>
>
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