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

Thu May 15 10:47:28 CDT 2014

Hello,

Thank you for your response.

Actually, I managed to solve a part of the problem; as I use Kamailio to
handle authentication, problem was that even though authentication went ok
through Kamailio, Asterisk refused to accept messages from Kamailio. That's
why Asterisk sent the 401. I think I had incorrect values in the realtime
sippeers table rows, and also I had to add values to deny and permit
fields, which in fact were in the wrong order. So no wonder I was having
problems with authentication! (and yes, I do know how digest authentication
works ;))

I fixed the deny values to 0.0.0.0/0.0.0.0 and permit value to Kamailio ip.

Even after this I had problems having Asterisk accept the authentications.
Asterisk cli was saying:
ERROR[20605]: chan_sip.c:30790 build_peer: Bad ACL entry in configuration
line 0 : kamailioip:5060

... that was because I had tried to define kamailio ip with port, as
Kamailio and Asterisk are on the same machine, but removing the port solved
that (not sure but I guess it is good I use 5060 for Kamailio and 5070 for
Asterisk instead of vice versa, perhaps this solution wouldn't work then).
Then I found that I had to add values to fields: nat (to force_rport) and
defaultip (to 0.0.0.0), and only after that I got Asterisk to see the
registered peers. So now everything looks ok from both Asterisk and
Kamailio when it comes to authentication.

I still can't get calls going though, in the asterisk cli I get 'Everyone
is busy/congested at this time', so I'm going to continue investigating
that. If you guys have good advice for me at this time I'll be most happy
to take them.

cheers,
Olli

2014-05-15 17:17 GMT+03:00 Leandro Dardini <ldardini at gmail.com>:

> 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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>
>
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