<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
</head>
<body>
<font face="monospace">Check<br>
<a class="moz-txt-link-freetext" href="https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/">https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/</a><br>
<a class="moz-txt-link-freetext" href="https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/">https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/</a><br>
<br>
Regards,<br>
</font>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="https://www.opensips-solutions.com">https://www.opensips-solutions.com</a>
<a class="moz-txt-link-freetext" href="https://www.siphub.com">https://www.siphub.com</a></pre>
<div class="moz-cite-prefix">On 5/24/23 1:00 AM, Dylan Cruz wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CA+3tVPSDppiTsRUZrqFK81NKLewdtWtAyxzUC3aFjboHBKQAbg@mail.gmail.com">
<meta http-equiv="content-type" content="text/html; charset=UTF-8">
<div dir="ltr">Still looking for possibly a template/example code
on this.
<div><br>
</div>
<div>I am setting a bounty of $150 for anyone willing to help.</div>
<div><br>
</div>
<div>You can reach out to me via E-Mail or phone at 407-999-0000</div>
<div><br>
</div>
<div>Thanks!</div>
</div>
<br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Mon, Mar 13, 2023 at
8:26 PM Dylan Cruz <<a href="mailto:dylan@regtelco.com"
moz-do-not-send="true">dylan@regtelco.com</a>> wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr">
<div style="color:rgb(0,0,0);font-family:"YS
Text",Arial,sans-serif;font-size:16px">I'd love a
sample OpenSIPS Config that would let me accomplish using
it as a transparent proxy to Asterisk running on the same
system. I found a few tutorials but found a lot of
conflicting information and outdated sources, Once I have
that I will have enough to work on to do what I want...
Basically I would like OpenSIPS to sit between the outside
world and Asterisk, Incoming & Outgoing would both
transparently be proxied through it. OpenSIPS would be
running on port 5060 & Asterisk would be running on
port 5090, So for example to register to a SIP Trunk from
a VoIP provider my Asterisk sip.conf would look like this:
(I know chan_sip is deprecated...)</div>
<div style="color:rgb(0,0,0);font-family:"YS
Text",Arial,sans-serif;font-size:16px"> </div>
<div style="color:rgb(0,0,0);font-family:"YS
Text",Arial,sans-serif;font-size:16px">
<div><strong>[general]</strong></div>
<div><strong>nat=no</strong></div>
<div><span style="color:rgb(0,0,255)"><strong>bindport=5090</strong></span></div>
<div><strong><span style="color:rgb(0,0,255)">outboundproxy=<a
href="http://127.0.0.1:5060" target="_blank"
moz-do-not-send="true">127.0.0.1:5060</a></span><span
style="color:rgb(238,130,238)"> </span>; Route
everything through OpenSIPS</strong></div>
<div><strong>tos_sip=cs3</strong></div>
<div><strong>tos_audio=ef</strong></div>
<div><strong>trustrpid=yes</strong></div>
<div><strong>canreinvite=yes</strong></div>
<div><strong>directrtpsetup=yes</strong></div>
<div><strong>allowguest=no</strong></div>
<div><strong>allowoverlap=yes</strong></div>
<div><strong>srvlookup=yes</strong></div>
<div><strong>disallow=all</strong></div>
<div><strong>allow=ulaw</strong></div>
<div> </div>
<div><span style="color:rgb(0,0,205)"><strong>[inbound-pstn]</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>type=peer</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>host=191.122.31.32</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>insecure=invite,port</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>qualify=yes</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>context=from-inbound</strong></span></div>
<div> </div>
<div><span style="color:rgb(0,0,205)"><strong>[outbound-pstn]</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>type=peer</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>host=191.122.31.33</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>insecure=invite,port</strong></span></div>
<div><span style="color:rgb(0,0,205)"><strong>qualify=yes</strong></span></div>
<div> </div>
<div>
<div>I would then be able to talk to both of those
trunks from Asterisk and have inbound & outbound
calls working all the way through to the VoIP
provider.</div>
<div> </div>
<div>My purpose for wanting to do this is I want to play
around with the SIP-I module in OpenSIPS to interwork
ISUP IAM fields by breaking them out into SIP Headers
that I can then manipulate easily in Asterisk.</div>
<div><br>
</div>
<div>
<div>Full disclosure: I am a complete OpenSIPS noob!
This would be my first OpenSIPS project, I am very
impressed with its capabilities and by having a
little sample config it would allow me to
experiment and start my journey of getting my feet
wet with it!</div>
<div><br>
</div>
</div>
<div>Thanks in advance!</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<pre class="moz-quote-pre" wrap="">_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
</pre>
</blockquote>
<br>
</body>
</html>