<div dir="ltr"><div><div><div><div><div><div>Daniel,<br><br></div>Thank you very much for your answer! Turning up verbosity indeed brought up the following message:<br><br>chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"49450386473" <<a href="mailto:sip%3Axxxxxxxx@198.23.139.21">sip:xxxxxxxx@198.23.139.21</a>>;tag=as72a4117a'<br>
-- SIP/1001-00000006 is circuit-busy<br><br></div>... which in combination with this message about FreePBX integration on the mailing-list<br><br><a href="http://lists.sip-router.org/pipermail/sr-users/2013-May/077985.html">http://lists.sip-router.org/pipermail/sr-users/2013-May/077985.html</a><br>
<br></div>helped open my eyes that FreePBX is using different tables which in turn leads to the above authentication problem.<br><br></div>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. <br>
<br>There has been another project at <br><br><a href="http://www.ictinnovations.com/content/kamailio-elastix-installation-and-integeration">http://www.ictinnovations.com/content/kamailio-elastix-installation-and-integeration</a><br>
<br></div>giving it a shot, but their approach seems to be overkill and hard to maintain compared with your easy and elegant solution.<br><br></div>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?<br>
<div><div><div><div><div><div><div><br></div><div>Best Regards,<br>Michael<br></div></div></div></div></div></div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, May 29, 2013 at 10:43 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Hello,<br>
<br>
maybe you can run asterisk in debug mode and get more hints from the
log messages.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 5/28/13 3:35 PM, Michael Leuker
wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">
<div>
<div>
<div>
<div>
<div>
<div>
<div>Hi everybody,<br>
<br>
</div>
I've been following Daniel's excellent tutorial at<br>
<br>
<a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb" target="_blank">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a><br>
<br>
</div>
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<br>
<br>
== Everyone is busy/congested at this time (1:0/0/1)<br>
<br>
</div>
and plays the according VM message. "sip show peers"
shows the extension as online / the registration
correctly passed along by Kamailio:<br>
<br>
1001/1001 198.23.139.21 D A
5060 OK (1 ms)<br>
<br>
</div>
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. <br>
<br>
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).<br>
<br>
</div>
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).<br>
<br>
</div>
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<br>
<br>
</div>
Best Regards,<br>
Michael<br>
</div>
<br>
<fieldset></fieldset>
<br>
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<br>
<pre cols="72">--
Daniel-Constantin Mierla - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* <a href="http://asipto.com/u/katu" target="_blank">http://asipto.com/u/katu</a> *</pre>
</font></span></div>
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