Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?
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 ohjelmistoarkkite...@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 ohjelmistoarkkite...@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@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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Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?
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 ldard...@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 ohjelmistoarkkite...@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 ohjelmistoarkkite...@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@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?
Re: [asterisk-users] Realtime integration: Unregistered clients showing as registered?
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 ohjelmistoarkkite...@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@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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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@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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
