[asterisk-users] Realtime integration: Unregistered clients showing as registered?

[Leandro Dardini]

It is the way it works. First the phone sends a REGISTER without any
password. Asterisk answers with a "Unauthorized" and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.

Leandro


2014-05-14 13:12 GMT+02:00 Olli Heiskanen <ohjelmistoarkkitehti at gmail.com>:

>
> Hello,
>
> After a small break from working on this, I got the idea of tcpdumping the
> correct ports. What I see is REGISTER messages from Kamailio port to
> Asterisk, which are replied with 401 Unauthorized. Why is this happening?
> In my sippeers table the secret field has no value (tried both NULL and
> empty string) and the added field sippasswd has the correct password for
> the user.
>
> The above might be the cause of my problem, would anyone be able to advice
> me to get to correct behaviour? Now Kamailio sees the clients as
> registered, which would be wrong if Asterisk doesn't.
>
> cheers,
> Olli
>
>
>
> 2014-04-24 11:27 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti at gmail.com>
> :
>
>
>> Hello all,
>>
>> I've been testing a Kamailio Asterisk Realtime integration, and found a
>> strange situation.
>>
>> My problem is that when using the integration, everything seems ok but
>> Asterisk does not see the clients as registered. Kamailio and the clients
>> report registered clients. Also calls fail.
>>
>> In Asterisk cli sip show peers shows nothing but for example realtime
>> load sipusers name 660 shows the user data. Field regseconds has a value
>> and fullcontact has value 'sip:660 at 127.0.0.1:5060' (kamailio ip:port as
>> they are on the same machine).
>>
>> I have a very simple dialplan:
>>
>> [general]
>>
>> [default]
>> exten => _XXX,1,NoOp(general : Dialed ${EXTEN})
>>  same => n,Dial(SIP/${EXTEN},3600,rt)
>>  same => n,Hangup
>>
>>
>> Here's more on my problem and background to it, guys on the Kamailio list
>> helped out but looks like I need to check my Asterisk configuration.
>> https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html
>>
>> My goal is to have all clients in the asterisk database, asterisk (one at
>> this point, several later) handling the calls and Kamailio as proxy. In
>> Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one
>> domain 'testers.com'.
>>
>> I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on
>> the same rental virtual server. Clients are in my home network behind nat.
>> In MySQL I have database asterisk with table sippeers, where I have
>> clients added like this:
>> INSERT INTO sippeers
>> (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type)
>> VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com
>> ','660','friend');
>>
>> In this message there are some outputs and a sip trace of a register:
>> https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html
>>
>> What I don't know is how to configure sip.conf, so far I've just been
>> making guesses based on online examples and documentation.
>> My current sip.conf looks like this:
>>
>> [general]
>> bindport = 5070
>> bindaddr = 127.0.0.1
>> tcpbindaddr = 127.0.0.1:5070
>> tcpenable = no
>> limitonpeers = yes
>> ;rtcachefriends = yes
>> tos_sip=cs3
>> tos_audio=ef
>> realm = testers.com
>>
>> I've tried defining realm and domain values, but I lack proper
>> understanding of those. Can you guys help me out? Are there any other
>> configurations I need to check?
>>
>> Respectfully,
>> Olli
>>
>>
>>
>
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