Kamailio 4.0 & Asterisk realtime (Failed to import bind_ob, auth: qop set, but nonce-count ...)
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
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:
- 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.
- 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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla miconda@gmail.com:
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!
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+With... 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@edvina.net wrote:
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla miconda@gmail.com:
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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+With... 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@edvina.net wrote:
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla miconda@gmail.com:
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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+With... 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@edvina.net wrote:
7 jun 2013 kl. 11:20 skrev Daniel-Constantin Mierla miconda@gmail.com:
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Ingenieurbüro Martin -
Olle E. Johansson -
Thomas Martin