Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)
5 Jun
2013
5 Jun
'13
10:59 p.m.
Hello,
being new to Kamailio, I have been closely following Daniel's tutorial "Kamailio
4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database",
hoping to be able to setup a basic SIP server with voicemail-boxes attached to the the
accounts. (Eventually, UACs that live behind either port-restricted or
symmetric NATs should be supported (i.e. experience two way audio), so, I guess,
eventually "WITH_NAT" needs to be defined and rtpproxy needs to be
installed as described in the "Run your own Skype-like service in less than one
hour" tutorial... ?)
Anyway, I am encountering a couple of difficulties in an early stage and hope for some
insights form the experts:
1) Even though launching Kamailio with "/etc/init.d/kamailio start" seems to
succeed ...
root@jm1:~# /etc/init.d/kamailio start
Starting Kamailio:
loading modules under /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/
Listening on
udp: 127.0.0.1:5060
udp: 178.254.20.156:5060
tcp: 127.0.0.1:5060
tcp: 178.254.20.156:5060
Aliases:
tcp: rv1192.1blu.de:5060
tcp: localhost:5060
udp: rv1192.1blu.de:5060
udp: localhost:5060
*: jingle-me.org:*
kamailio started.
root@jm1:~#
... SIP Clients (I have tried several) cannot register and receive the response: 401
Unauthorised.
cat /var/log/syslog yields:
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr [../outbound/api.h:49]: Failed to
import bind_ob
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr [rr_mod.c:159]: outbound module not
available
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: usrloc [hslot.c:53]: locks array size
512
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: auth [auth_mod.c:350]: auth: qop set,
but nonce-count (nc_enabled) support disabled
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:179]: INFO:
udp_init: SO_RCVBUF is initially 229376
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:230]: INFO:
udp_init: SO_RCVBUF is finally 262142
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:179]: INFO:
udp_init: SO_RCVBUF is initially 229376
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core> [udp_server.c:230]: INFO:
udp_init: SO_RCVBUF is finally 262142
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2200]: INFO: ctl [io_listener.c:225]:
io_listen_loop: using epoll_lt io watch method (config)
I suspect that the "INFO: auth [auth_mod.c:350]: auth: qop set, but nonce-count
(nc_enabled) support disabled" hints to the problem, but have no clue how to fix
that.
2) Kamailio does not start on boot-up, even though "/etc/init.d/kamailio start
(stop/restart/status)" work(s), and /etc/default/kamailio looks like this:
RUN_KAMAILIO=yes
USER=kamailio
GROUP=kamailio
SHM_MEMORY=64
PKG_MEMORY=4
SSD_SUID=no
DUMP_CORE=no
Right after a re-boot, root@jm1:~# chkconfig kamailio yields: "kamailio on",
however, root@jm1:~#/etc/init.d/kamailio status yields: "Status of Kamailio: kamailio
is not running." at this time.
Any hints are appreciated!
Best regards,
-Thomas
7 Jun
7 Jun
11:20 a.m.
New subject: Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)
Hello,
first, as pointed in other related discussions in this mailing list, it
might be better to use a different approach if you start everything from
scratch. That will be doing all signaling handling in kamailio and use
asterisk only as media server. Practically all user data is in kamailio,
routing to asterisk only when needed for media services. The tutorial is
more targeting existing asterisk deployments. Nevertheless, see more
comments inline.
On 6/5/13 10:59 PM, Thomas Martin wrote:
Hello,
being new to Kamailio, I have been closely following Daniel's tutorial
"Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using
Asterisk Database",
hoping to be able to setup a basic SIP server with voicemail-boxes
attached to the the accounts. (Eventually, UACs that live behind
either port-restricted or
symmetric NATs should be supported (i.e. experience two way audio),
so, I guess, eventually "WITH_NAT" needs to be defined and rtpproxy
needs to be
installed as described in the "Run your own Skype-like service in less
than one hour" tutorial... ?)
Anyway, I am encountering a couple of difficulties in an early stage
and hope for some insights form the experts:
1) Even though launching Kamailio with "/etc/init.d/kamailio start"
seems to succeed ...
root@jm1:~# /etc/init.d/kamailio start
Starting Kamailio:
loading modules under
/usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/
Listening on
udp: 127.0.0.1:5060
udp: 178.254.20.156:5060
tcp: 127.0.0.1:5060
tcp: 178.254.20.156:5060
Aliases:
tcp: rv1192.1blu.de <http://rv1192.1blu.de>:5060
tcp: localhost:5060
udp: rv1192.1blu.de <http://rv1192.1blu.de>:5060
udp: localhost:5060
*: jingle-me.org <http://jingle-me.org>:*
kamailio started.
root@jm1:~#
... SIP Clients (I have tried several) cannot register and receive the
response: 401 Unauthorised.
cat /var/log/syslog yields:
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr
[../outbound/api.h:49]: *Failed to import bind_ob*
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: rr [rr_mod.c:159]:
outbound module not available
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: usrloc
[hslot.c:53]: locks array size 512
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: auth
[auth_mod.c:350]: auth: qop set, but nonce-count (nc_enabled) *support
disabled*
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core>
[udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core>
[udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core>
[udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 229376
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2180]: INFO: <core>
[udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142
Jun 5 21:59:01 jm1 /usr/sbin/kamailio[2200]: INFO: ctl
[io_listener.c:225]: io_listen_loop: using epoll_lt io watch method
(config)
I suspect that the "INFO: auth [auth_mod.c:350]: auth: qop set, but
nonce-count (nc_enabled) *support disabled*" hints to the problem, but
have no clue how to fix that.
It should go fine with this message (to try to get
rid of it, paste here
the module parameters for auth and auth_db modules). Anyhow, the logs
you provide are from startup. You have to get the logs when the phone
attempts to register. Set debug=3 in your config, restart and then
register with a phone, you should see lots of log messages. Paste them
to mailing list if you cannot sort out the issue by yourself.
2) Kamailio does _not_ start on boot-up, even though
"/etc/init.d/kamailio start (stop/restart/status)" work(s), and
/etc/default/kamailio looks like this:
RUN_KAMAILIO=yes
USER=kamailio
GROUP=kamailio
SHM_MEMORY=64
PKG_MEMORY=4
SSD_SUID=no
DUMP_CORE=no
Right after a re-boot, root@jm1:~# chkconfig kamailio yields:
"kamailio on",
however, root@jm1:~#/etc/init.d/kamailio status yields: "Status of
Kamailio: kamailio is not running." at this time.
Can you do "ps auxw | grep kamailio" and see if kamailio processes are
listed?
Cheers,
Daniel
Any hints are appreciated!
Best regards,
-Thomas
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
11:33 a.m.
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Hello,
first, as pointed in other related discussions in this mailing list, it might be better
to use a different approach if you start everything from scratch. That will be doing all
signaling handling in kamailio and use asterisk only as media server. Practically all user
data is in kamailio, routing to asterisk only when needed for media services. The tutorial
is more targeting existing asterisk deployments. Nevertheless, see more comments inline.
Here's a presentation from Astricon 2010 where I discuss multiple types of
integration between Asterisk and Kamailio. The one in the tutorial
is, as Daniel says, focused on limited impact on an existing Asterisk installation and
it's not one I recommend if you start from scratch with
a new architecture.
Read it through to get a view of a couple of different approaches:
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
/O
-----
Edvina SIP Masterclass in Malaga, Spain, July 2013
Learn more about Kamailio and SIP!
http://edvina.net/blog/2013/01/sipmaster-malaga-2013/
Register now!
9:18 p.m.
Hello,
thanks to your responses.
In the meantime, I have read Olle's slides a few times trying to understand the
ramifications of the different approaches outlined (I am new to kamailio and pretty new to
asterisk too). Routing calls to asterisk only when needed for media services and keeping
all user data in kamailio, seems to be the fitting approach for my desires.
It seems as if the
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+Wit…
explains how to setup a system like that. - However, the article refers to older versions
of both, asterisk & kamailio.
Would you still recommend to follow those same instructions but use the current releases
(11.4 & 4.0.1)? - Or can you push me into a better direction ?
Again, thank you very much for your help!
Best regards,
-Thomas
ps: Being new to this mailing list, I am unfortunately unaware of topics that might
already have been exhaustedly covered - I apologise. - Also, I decided to reinstall
everything and start from scratch - at this time don't want to bother anybody with the
logs that just document previously failing attempts.
On Jun 7, 2013, at 11:33 , "Olle E. Johansson" <oej(a)edvina.net> wrote:
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Hello,
first, as pointed in other related discussions in this mailing list, it might be better
to use a different approach if you start everything from scratch. That will be doing all
signaling handling in kamailio and use asterisk only as media server. Practically all user
data is in kamailio, routing to asterisk only when needed for media services. The tutorial
is more targeting existing asterisk deployments. Nevertheless, see more comments inline.
Here's a presentation from Astricon 2010 where I discuss multiple types of
integration between Asterisk and Kamailio. The one in the tutorial
is, as Daniel says, focused on limited impact on an existing Asterisk installation and
it's not one I recommend if you start from scratch with
a new architecture.
Read it through to get a view of a couple of different approaches:
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
/O
-----
Edvina SIP Masterclass in Malaga, Spain, July 2013
Learn more about Kamailio and SIP!
http://edvina.net/blog/2013/01/sipmaster-malaga-2013/
Register now!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
9:23 p.m.
Hello,
thanks to your responses.
In the meantime, I have read Olle's slides a few times trying to understand the
ramifications of the different approaches outlined (I am new to kamailio and pretty new to
asterisk too). Routing calls to asterisk only when needed for media services and keeping
all user data in kamailio, seems to be the fitting approach for my desires.
It seems as if the
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+Wit…
explains how to setup a system like that. - However, the article refers to older versions
of both, asterisk & kamailio.
Would you still recommend to follow those same instructions but use the current releases
(11.4 & 4.0.1)? - Or can you push me into a better direction ?
Again, thank you very much for your help!
Best regards,
-Thomas
ps: Being new to this mailing list, I am unfortunately unaware of topics that might
already have been exhaustedly covered - I apologise. - Also, I decided to reinstall
everything and start from scratch - at this time don't want to bother anybody with the
logs that just document previously failing attempts.
On Jun 7, 2013, at 11:33 , "Olle E. Johansson" <oej(a)edvina.net> wrote:
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Hello,
first, as pointed in other related discussions in this mailing list, it might be better
to use a different approach if you start everything from scratch. That will be doing all
signaling handling in kamailio and use asterisk only as media server. Practically all user
data is in kamailio, routing to asterisk only when needed for media services. The tutorial
is more targeting existing asterisk deployments. Nevertheless, see more comments inline.
Here's a presentation from Astricon 2010 where I discuss multiple types of
integration between Asterisk and Kamailio. The one in the tutorial
is, as Daniel says, focused on limited impact on an existing Asterisk installation and
it's not one I recommend if you start from scratch with
a new architecture.
Read it through to get a view of a couple of different approaches:
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
/O
-----
Edvina SIP Masterclass in Malaga, Spain, July 2013
Learn more about Kamailio and SIP!
http://edvina.net/blog/2013/01/sipmaster-malaga-2013/
Register now!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
10 Jun
10 Jun
9:07 a.m.
New subject: Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)
Hello,
On 6/7/13 9:23 PM, Thomas Martin wrote:
Hello,
thanks to your responses.
In the meantime, I have read Olle's slides a few times trying to understand the
ramifications of the different approaches outlined (I am new to kamailio and pretty new to
asterisk too). Routing calls to asterisk only when needed for media services and keeping
all user data in kamailio, seems to be the fitting approach for my desires.
It seems as if the
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+Wit…
explains how to setup a system like that. - However, the article refers to older versions
of both, asterisk & kamailio.
Would you still recommend to follow those same instructions but use the current releases
(11.4 & 4.0.1)? - Or can you push me into a better direction ?
that wiki page,
even old, still gives good overview of how to do it, it
may require to do some adjustments to match database structures and
config files from the latest versions of asterisk and kamailio.
Cheers,
Daniel
Again, thank you very much for your help!
Best regards,
-Thomas
ps: Being new to this mailing list, I am unfortunately unaware of topics that might
already have been exhaustedly covered - I apologise. - Also, I decided to reinstall
everything and start from scratch - at this time don't want to bother anybody with the
logs that just document previously failing attempts.
On Jun 7, 2013, at 11:33 , "Olle E. Johansson" <oej(a)edvina.net> wrote:
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
7 jun 2013 kl. 11:20 skrev Daniel-Constantin
Mierla <miconda(a)gmail.com>om>:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Hello,
first, as pointed in other related discussions in this mailing list, it might be better
to use a different approach if you start everything from scratch. That will be doing all
signaling handling in kamailio and use asterisk only as media server. Practically all user
data is in kamailio, routing to asterisk only when needed for media services. The tutorial
is more targeting existing asterisk deployments. Nevertheless, see more comments inline.
Here's a presentation from Astricon 2010 where I discuss multiple types of
integration between Asterisk and Kamailio. The one in the tutorial
is, as Daniel says, focused on limited impact on an existing Asterisk installation and
it's not one I recommend if you start from scratch with
a new architecture.
Read it through to get a view of a couple of different approaches:
http://www.slideshare.net/oej/astricon-2010-scaling-asterisk-installations
/O
-----
Edvina SIP Masterclass in Malaga, Spain, July 2013
Learn more about Kamailio and SIP!
http://edvina.net/blog/2013/01/sipmaster-malaga-2013/
Register now!
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4191
days inactive
4196
days old
5 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
Ingenieurbüro Martin -
Olle E. Johansson -
Thomas Martin