<div dir="ltr">Hi Bogdan<div><br></div><div style>Thanks for the suggestion. If I understand you right, I should have rule to try the subscriber locally first before sending it to the LB.</div><div style>I'll try and figure it out. I'm still trying to wrap my head around the routing concept in Opensips, coming from Asterisk diaplan syntax background.</div>
<div style><br></div><div style>Regards</div><div style>Stephen</div><div style><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Jun 28, 2013 at 11:18 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div text="#000000" bgcolor="#ffffff">
Hello Stephen,<br>
<br>
When receiving a call in opensips you need (by looking at the dialed
number) to make difference between subscribers (extensions) and call
to be sent to LB. Can you do that ?<br>
<br>
Regards,<br>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre><div><div class="h5">
<br>
On 06/28/2013 09:36 AM, Stephen Vigus wrote:
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">Hi guys
<div><br>
</div>
<div>First time posting to the list.</div>
<div><br>
</div>
<div>I am investigating the use of Opensips to load
balance between Asterisk boxes. I have a running testing
environment, where Opensips successfully load balances between
the Asterisk boxes. However, after a few days of reading and
struggling, I came to a dead end with a specific test.</div>
<div><br>
</div>
<div>I have my extensions registered to Opensips and
want them to be able to phone each other, but I cannot seem to
get this working. Every time I attempt to dial another
extension, I see the dialplan triggered on one of the Asterisk
boxes and then it fails.</div>
<div>Calls coming in over a trunk are load balanced
fine and reaches an IVR I configured on both Asterisk boxes.</div>
<div><br>
</div>
<div>What am I missing or doing wrong?</div>
<div><br>
</div>
<div>
Below are my routing in opensips.cfg:</div>
<div>
<div>route{</div>
<div><br>
</div>
<div> if (!mf_process_maxfwd_header("10")) {</div>
<div> sl_send_reply("483","Too Many Hops");</div>
<div> exit;</div>
<div> }</div>
<div><br>
</div>
<div> if (!has_totag()) {</div>
<div> # initial request</div>
<div> record_route();</div>
<div> }</div>
<div> else {</div>
<div> # sequential request - obey the indicated
route</div>
<div> lookup("location");</div>
<div> loose_route();</div>
<div> t_relay();</div>
<div> exit;</div>
<div> }</div>
<div><br>
</div>
<div> # handle cancel and re-transmissions</div>
<div> if ( is_method("CANCEL") ) {</div>
<div> if ( t_check_trans() )</div>
<div> t_relay();</div>
<div> exit;</div>
<div> }</div>
<div><br>
</div>
<div> # record routing</div>
<div> if (!is_method("REGISTER|MESSAGE"))</div>
<div> record_route();</div>
<div><br>
</div>
<div> if (is_method("REGISTER"))</div>
<div> {</div>
<div> if (!save("location"))</div>
<div> sl_reply_error();</div>
<div> exit;</div>
<div>
}</div>
<div><br>
</div>
<div><br>
</div>
<div> # from now on we have only the initial requests</div>
<div><br>
</div>
<div> # select the node that'll handle the call (load
balanced)</div>
<div> # the method used is different for
invite/register requests</div>
<div> # unknown methods are rejected here</div>
<div> if (is_method("INVITE")) {</div>
<div> if (!load_balance("1","pstn","1")) {</div>
<div> send_reply("503","Service
Unavailable");</div>
<div> exit;</div>
<div> }</div>
<div> }</div>
<div><br>
</div>
<div> else {</div>
<div> send_reply("405","Method Not Allowed");</div>
<div> exit;</div>
<div> }</div>
<div><br>
</div>
<div> # route the request</div>
<div> if (!t_relay()) {</div>
<div> sl_reply_error();</div>
<div> }</div>
<div>}</div>
<div><br>
</div>
<div><br>
</div>
<div>
Please excuse me if this comes off as daft, this week was my
first look into Opensips.</div>
<div><br>
</div>
<div>Regards</div>
<div>Stephen</div>
</div>
</div>
</div></div><pre><fieldset></fieldset>
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</pre>
</blockquote>
</div>
</blockquote></div><br></div>