<div dir="ltr"><div class="gmail_extra"><div class="gmail_quote">On 26 April 2014 00:29, Alex Villacís Lasso <span dir="ltr"><<a href="mailto:a_villacis@palosanto.com" target="_blank">a_villacis@palosanto.com</a>></span> wrote:<br>
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<div style="font-family:-moz-fixed;font-size:14px" lang="x-western">I am currently preparing a
kamailio-asterisk combination. The asterisk installation uses
realtime for SIP. The kamailio configuration was based on the
reference at <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>
but has been heavily modified. Currently asterisk runs on
localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore,
all of the SIP traffic appears to come from localhost, from the
point of view of asterisk.
<br>
<br>
Currently I have a model on which internal SIP phones get
identified by the authentication username, and then the contact
names at From: and To: get massaged to incorporate the SIP domain,
in order to emulate multiple-domain support. The 'sip' table in
Asterisk defines all such contacts as SIP accounts of the form
<a href="http://name_domain.com" target="_blank">name_domain.com</a>, and the SIP phones are configured to use 'name'
as authentication username for domain '<a href="http://domain.com" target="_blank">domain.com</a>'. However, SIP
providers that register on the server with authentication names
are left with their original names, since in the model, SIP trunks
are available to all domains.
<br>
<br>
Now I have to add support for SIP providers which are to be
authorized on the basis of IP only. Apparently, the kamailio
module permissions.so (WITH_IPAUTH) is made for just this purpose,
so I enabled it. After authentication, I need to route the INVITE
to asterisk, and asterisk must somehow match the account for the
SIP trunk from the available information on the INVITE request.
<br>
<br></div></div></blockquote><div><br></div><div><br></div><div>What I have done in a similar situation is to use force_send_socket in Kamailio when sending INVITEs from your trusted host (your trunks) so that it is coming in to Asterisk from a different port (say 5070), and then in your Asterisk sip.conf settings create a new peer for this like so:</div>
<div><br></div><div>[peer-incoming]</div><div><div style="font-family:arial,sans-serif;font-size:13px">context=peercontext</div><div style="font-family:arial,sans-serif;font-size:13px">type=peer</div><div style="font-family:arial,sans-serif;font-size:13px">
host=127.0.0.1</div></div><div style="font-family:arial,sans-serif;font-size:13px">port=5070</div><div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">
Now, when Asterisk receives an INVITE from <a href="http://127.0.0.1:5070">127.0.0.1:5070</a> it will match this peer, whereas the rest, coming from <a href="http://127.0.0.1:5060">127.0.0.1:5060</a>, will match your other subscribers.</div>
<div style="font-family:arial,sans-serif;font-size:13px"><br></div><div style="font-family:arial,sans-serif;font-size:13px">Here is a bit of the Kamailio config:</div><div style="font-family:arial,sans-serif;font-size:13px">
<br></div><div style="font-family:arial,sans-serif;font-size:13px"><div>if (is_method("INVITE"))</div><div> {</div><div> # If call is coming from a trusted source (Trunk/PSTN) then we send it to Asterisk from port 5070</div>
<div> # so that Asterisk knows this is not coming from a subscriber. The peer in Asterisk needs to be set with port=5070</div><div> # as well as the host=<ip address></div><div> if (allow_trusted())</div>
<div> {</div><div> xlog("L_INFO","Inbound to Asterisk from Trusted Source IP $si, Caller: $fU, Callee: $rU with Call-ID $hdr(Call-ID)");</div><div> force_send_socket(<a href="http://127.0.0.1:5070">127.0.0.1:5070</a>);</div>
<div> } else {</div><div> # This is a call from a registered subscriber.</div><div> xlog("L_INFO","Inbound to Asterisk from $fU to $rU with Call-ID $hdr(Call-ID)");</div><div>
} </div><div> }</div><div> route(RELAY);</div><div> exit;</div><div>}</div><div><br></div></div><div>NOTE: Kamailio must be set to listen on <a href="http://127.0.0.1:5070">127.0.0.1:5070</a> as well as your usual ports for this to work! Also, your SIP Trunk trusted peers need to be in the Kamailio trusted table, or explicitly test for the src_ip rather than use allow_trusted().</div>
<div><br></div><div>Hope this helps.</div><div><br></div><div>-Barry Flanagan</div><div><br></div><div><br></div></div></div></div>