asterisk users - Apr 2014 - Realtime integration: Unregistered clients showing as registered?

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 at 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 at 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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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 at 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 at 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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