Hi everybody,
I've been following Daniel's excellent tutorial at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and managed to get registrations and outbound dialing to run. The problem is that whenever I try to call a local extension (directly or via trunk), CLI reports
== Everyone is busy/congested at this time (1:0/0/1)
and plays the according VM message. "sip show peers" shows the extension as online / the registration correctly passed along by Kamailio:
1001/1001 198.23.139.21 D A 5060 OK (1 ms)
198.23.139.21 is the server IP and used for both Kamailio (5060) and Asterisk (5080). Asterisk only allows for UDP, Kamailio for for UDP and TCP connections.
The server is connected directly to the internet and the only NAT is on the client side. I have disabled NAT support both in Asterisk and Kamailio because the clients support ICE and can connect using TURN to the Asterisk echo-test just fine. Enabling NAT support (with and without rtpproxy) in Kamailio doesn't solve the above problem (and doesn't seem to be able to traverse all NAT situations in any case).
To be clear: The Asterisk configuration itself is working flawlessly. I don't encounter the problem at all if I connect directly to the Asterisk server, either in its original, standalone configuration or on port 5080 (after disabling the ACL allowing only the Kamailio IP and re-enabling the password).
Please let me know if you need specific logs (I wouldn't want to needlessly clutter the original description with useless information) to take a closer look at the problem. If this is a known issue, a friendly pointer would be much appreciated as well! I've searched for hours and hours and never seemed to find anything helpful. Have a great day and with
Best Regards, Michael
Hello,
maybe you can run asterisk in debug mode and get more hints from the log messages.
Cheers, Daniel
On 5/28/13 3:35 PM, Michael Leuker wrote:
Hi everybody,
I've been following Daniel's excellent tutorial at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and managed to get registrations and outbound dialing to run. The problem is that whenever I try to call a local extension (directly or via trunk), CLI reports
== Everyone is busy/congested at this time (1:0/0/1)
and plays the according VM message. "sip show peers" shows the extension as online / the registration correctly passed along by Kamailio:
1001/1001 198.23.139.21 D A 5060 OK (1 ms)
198.23.139.21 is the server IP and used for both Kamailio (5060) and Asterisk (5080). Asterisk only allows for UDP, Kamailio for for UDP and TCP connections.
The server is connected directly to the internet and the only NAT is on the client side. I have disabled NAT support both in Asterisk and Kamailio because the clients support ICE and can connect using TURN to the Asterisk echo-test just fine. Enabling NAT support (with and without rtpproxy) in Kamailio doesn't solve the above problem (and doesn't seem to be able to traverse all NAT situations in any case).
To be clear: The Asterisk configuration itself is working flawlessly. I don't encounter the problem at all if I connect directly to the Asterisk server, either in its original, standalone configuration or on port 5080 (after disabling the ACL allowing only the Kamailio IP and re-enabling the password).
Please let me know if you need specific logs (I wouldn't want to needlessly clutter the original description with useless information) to take a closer look at the problem. If this is a known issue, a friendly pointer would be much appreciated as well! I've searched for hours and hours and never seemed to find anything helpful. Have a great day and with
Best Regards, Michael
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,
Thank you very much for your answer! Turning up verbosity indeed brought up the following message:
chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"49450386473" sip:xxxxxxxx@198.23.139.21;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy
... which in combination with this message about FreePBX integration on the mailing-list
http://lists.sip-router.org/pipermail/sr-users/2013-May/077985.html
helped open my eyes that FreePBX is using different tables which in turn leads to the above authentication problem.
I'll take a look at the configuration this weekend, but am far from sure that I can make the necessary changes. For what it's worth, FreePBX is an extremely popular Asterisk GUI that does a fantastic job making Asterisk accessble for the rest of us.
There has been another project at
http://www.ictinnovations.com/content/kamailio-elastix-installation-and-inte...
giving it a shot, but their approach seems to be overkill and hard to maintain compared with your easy and elegant solution.
Long story short: I could easily follow a tutorial, but will probably stumble in the dark for a long time with uncertain results if giving this a shot on my own. It's only a hobby for me, but I'd gladly pay for some help adapting the database settings, preferably in a way that leaves the FreePBX structure untouched. Anybody willing to give it a shot?
Best Regards, Michael
On Wed, May 29, 2013 at 10:43 AM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
maybe you can run asterisk in debug mode and get more hints from the log messages.
Cheers, Daniel
On 5/28/13 3:35 PM, Michael Leuker wrote:
Hi everybody,I've been following Daniel's excellent tutorial at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and managed to get registrations and outbound dialing to run. The problem is that whenever I try to call a local extension (directly or via trunk), CLI reports
== Everyone is busy/congested at this time (1:0/0/1)
and plays the according VM message. "sip show peers" shows the extension as online / the registration correctly passed along by Kamailio:
1001/1001 198.23.139.21 D A 5060 OK (1 ms)
198.23.139.21 is the server IP and used for both Kamailio (5060) and Asterisk (5080). Asterisk only allows for UDP, Kamailio for for UDP and TCP connections.
The server is connected directly to the internet and the only NAT is on the client side. I have disabled NAT support both in Asterisk and Kamailio because the clients support ICE and can connect using TURN to the Asterisk echo-test just fine. Enabling NAT support (with and without rtpproxy) in Kamailio doesn't solve the above problem (and doesn't seem to be able to traverse all NAT situations in any case).
To be clear: The Asterisk configuration itself is working flawlessly. I don't encounter the problem at all if I connect directly to the Asterisk server, either in its original, standalone configuration or on port 5080 (after disabling the ACL allowing only the Kamailio IP and re-enabling the password).
Please let me know if you need specific logs (I wouldn't want to needlessly clutter the original description with useless information) to take a closer look at the problem. If this is a known issue, a friendly pointer would be much appreciated as well! I've searched for hours and hours and never seemed to find anything helpful. Have a great day and with
Best Regards, Michael
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
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 -
Michael Leuker