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Hello,<br>
<br>
using a proxy in front of the pbx has some side effects if your
phones are behind the nat. First, for registration, you have to
enable with Path in the proxy and the pbx has to support it. AFAIK,
Asterisk got just recently a patch for supporting Path.<br>
<br>
Then, the asterisk has to be also changed a bit, because it will
match all phone registrations coming from the same IP, which can
create troubles if you match by source IP (not sure if this is still
in latest Asterisks, but with some versions I run in such troubles,
but since I was not the one handling the Asterisk, I cannot tell
more).<br>
<br>
If you really need to stick to existing asterisk deployment, then
maybe looking at next tutorial will help a bit:<br>
<br>
<a class="moz-txt-link-freetext" href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a><br>
<br>
If you can redesign the platform, you may want to change the logic
so that kamailio takes care of registration and signaling and
asterisk of media related services. This is more scalable and easier
to handle in long term.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 8/1/13 12:44 AM, Rafael Carvallo
wrote:<br>
</div>
<blockquote
cite="mid:CABo7iV+qtP_jkiRZBYG5=QObG-CTxwkDV04Q49xW5qNqiYc8bg@mail.gmail.com"
type="cite">
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<div>
<div>
<div>
<div>
<div>
<div>Hey everyone, currently i have this configuration
set on kamailio:<br>
<br>
<style type="text/css">P { margin-bottom: 0.21cm; }</style>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"list_file", "/etc/kamailio/dispatcher.list")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"force_dst", 0) #forzado de la reescritura
direccion de
destino</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"flags", 2) #banderas de funcionamiento, 2
significa
"soporte para failover"</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"dst_avp", "$avp(dsdst)")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"grp_avp", "$avp(dsgrp)")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"cnt_avp", "$avp(dscnt)")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_ping_method", "OPTIONS")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_ping_interval", 5) #tiempo que transcurre
antes de
verificar nuevamente una salida inactiva</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_probing_threshhold", 5) #Numero de
intentos antes de
marcar una salida como inactiva</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_ping_reply_codes",
"class=2;code=403;code=488;class=3")</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_probing_mode", 1)</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_hash_expire", 3600)</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>modparam("dispatcher",
"ds_hash_initexpire", 60)</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>#loadmodule
"dispatcher.so"</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>#######
Routing Logic ########</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>#
Main SIP request routing logic</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>#
- processing of any incoming SIP request
starts with this route</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>#
- note: this is the same as route { ... }</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>request_route
{</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>if
(is_method("SUBSCRIBE")){</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>if
(src_ip == 192.168.2.1 | src_ip ==
192.168.2.2){</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>t_relay();</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>}</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>route(REGISTRAR);</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>}</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>if
(src_ip == 192.168.2.1 | src_ip ==
192.168.2.2){</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>t_relay();</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>}</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>else{</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>route(ASTERISK);</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>}</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>}</font></font></p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><br>
</p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>route[ASTERISK]{</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>ds_select_dst("1",
"8");</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>t_relay();</font></font></p>
<p style="margin-bottom:0cm"> <font face="Arial,
serif"><font>exit();</font></font></p>
<p style="margin-bottom:0cm"><font face="Arial,
serif"><font>}</font></font></p>
<br>
</div>
<div>Everything else is left with the same basic
configuration it had when i installed the software.<br>
</div>
<div><br>
</div>
I want to use it kinda of a sip router, so what
Kamailio does is just forward the packets betwen my
Asterisk boxes and the Sip Phones. Currently it seems
to work (partially) but i have doubts about if this is
correctly done (it's the first time using Kamailio and
i need this working withing a week at most).<br>
<br>
</div>
As you can see i use the module dispatcher for
failover/failback (this is the purpose of using
Kamailio, a failover/failback setup). One major problem
i've found with this setup is, if the phones are
currently connected and working with one of my 2
asterisk boxes and if that box fails, Kamailio starts
sending the traffic to the second box (as intended), but
the phones don't try to subscribe to the new asterisk
box, rather they just keep sending traffic (and
obviously kamailio forwarding it).<br>
<br>
</div>
Sometimes one of the phones subscribe to the new box, but
that's not always the case, the packets reach the new
Asterisk box, but since the phones aren't registered to
it, they can't make calls.<br>
<br>
</div>
Other times the behavior of the setup is rather weird, i can
call some extensions and some others i can not (even though
they are registered within the asterisk box) The traffic
gets to the Asterisk box (as shown in the asterisk logs) but
the call is shown as "service unavailable". I've checked a
lot of times the Asterisk setup and it seems to be fine so i
think it has something to do with Kamailio.<br>
<br>
</div>
I know this might be a really bad configuration file, but it's
been at most 3 days since i started using Kamailio which i
sometimes find kinda hard to understand and i really need this
working within a week.<br>
<br>
</div>
To summarize all i want is kamailio forwarding packets between
the currently active server and the phones so if it fails, then
the packets go to the second one, the phones must re-subscribe
to the new active server. <br>
</div>
<br>
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<br>
<pre wrap="">_______________________________________________
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
</pre>
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