Perfect! Everything's working now. <div><br></div><div>Thanks again Bogdan.<br><br><div class="gmail_quote">On Fri, Mar 16, 2012 at 9:24 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org">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 bgcolor="#ffffff" text="#000000">
Joshua,<br>
<br>
if you want use regex for src IP matching, better do:<br>
if ($si !~ "^10\..*"){}<br>
<br>
And for printing, use the xlog function - it can print variables
too.<br>
4. log("source_ip: $si\n");<br>
<br>
Regards,<br>
Bogdan<div><div class="h5"><br>
<br>
On 03/16/2012 04:02 PM, Joshua Nankin wrote:
<blockquote type="cite">Sorry about that - I have a Gmail keyboard shortcut
reflex :)
<div><br>
</div>
<div>So, I'd like to do a regex based check where the PBXes all
have addresses starting with 10.*. I did the following:</div>
<div><br>
</div>
<div>
<div> if (src_ip !~ "^10\..*"){</div>
<div> if (!load_balance("1","fax","1")) {</div>
<div>
sl_send_reply("503","Service Unavailable");</div>
<div> exit;</div>
<div> }</div>
<div> }</div>
</div>
<div> else {</div>
<div> # this is an outgoing call, just
let the packets go through (somehow)</div>
<div> }</div>
<div><br>
</div>
<div>But it looks like my regex is not working and incoming calls
get a busy signal. To make matters worse, I tried debugging
with:</div>
<div><br>
</div>
<div>1. log(src_ip)</div>
<div>2. log("source ip: " + src_ip)</div>
<div>3. log($si)</div>
<div>4. log("source_ip: $si")</div>
<div><br>
</div>
<div>None of those works (all produce config file syntax errors
except for 4 which just prints out the actual $si characters.
Why can't I do the above?</div>
<div><br>
</div>
<div>Finally, what do I need to do in the "else" block to let the
outgoing call packets go through? I tried placing an outgoing
call with out the load_balancing block, but my packets were not
relayed from opensips to the carrier.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Josh </div>
<div><br>
<div class="gmail_quote">On Thu, Mar 15, 2012 at 1:34 PM,
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:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#ffffff" text="#000000"> Hi Joshua,<br>
<br>
Please keep the list at CC - whatever we talk here, should
also be to the benefit of the entire opensips community ;)<br>
<br>
So, to detect where the call comes from, you simply check
the source at IP level:<br>
if (src_ip==11.22.33.44) {<br>
# call from PBXes<br>
} else {<br>
# call from carriers<br>
}<br>
<br>
Regards,<br>
Bogdan
<div>
<div><br>
<br>
On 03/15/2012 07:33 PM, Joshua Nankin wrote:
<blockquote type="cite">Perfect! Thanks Bogdan,
you're the man! I was having some trouble with
record_route_preset because I was including the port
in the IP string as well (I had seen others do this
in alias, so I just copied that in the
record_route_preset function and things still didn't
work).
<div> <br>
</div>
<div>One more question: I'm trying to use the same
opensips box as an outbound proxy. Now, since I
automatically load balance any INVITEs, when I
place an outgoing call, the call is routed right
back to one of my Asterisk machines. How would I
tell if the request is incoming (and then load
balance it) or outgoing (and then relay the
messages outside of the network).</div>
<div><br>
</div>
<div>I'm sure this involves creating another route,
but I'm a complete n00b when it comes to opensips
- I just started messing around with it this week.</div>
<div><br>
</div>
<div>Thanks again for your help!<br>
<br>
<div class="gmail_quote">On Thu, Mar 15, 2012 at
4:40 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:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div bgcolor="#ffffff" text="#000000"> Hi
Joshua,<br>
<br>
Ok, I see - your opensips is actually in a
private network.<br>
What you can do is to use instead of
record_route() the
record_route_preset("pub_IP") function (see
<a href="http://www.opensips.org/html/docs/modules/1.7.x/rr.html#id250439" target="_blank">http://www.opensips.org/html/docs/modules/1.7.x/rr.html#id250439</a>)
- this will make opensips to advertise in RR
the public IP. Additionally set "alias =
pub_IP" in your cfg (see <a href="http://www.opensips.org/Resources/DocsCoreFcn17#toc26" target="_blank">http://www.opensips.org/Resources/DocsCoreFcn17#toc26</a>),
so that opensips will recognize its own pub
IP in the Route hdr (in ACK)<br>
<br>
Regards,<br>
Bogdan
<div><br>
<br>
On 03/14/2012 10:54 PM, Joshua Nankin
wrote: </div>
<blockquote type="cite">
<div>Do I need to specify the public IPs
of my asterisk boxes in the load
balancing table?
<div><br>
</div>
<div>Here's is what my carrier told me:</div>
<div><br>
</div>
<div><span>After looking into this issue
further, it appears you are not
including the Record-route header
for your public IP address. The
reason the call is failing currently
is because you have listed your
internal IP address for us to send
the call too, however since we do
not have access to your internal IP
address, we are unable to send this
to you. You can see this in the 200
OK that you send to us, where you
start with your internal IP, and go
straight to our IP.</span></div>
<div><font color="#500050" face="arial,
sans-serif"><br>
</font></div>
<div><font color="#500050" face="arial,
sans-serif">You can find a packet
captures for the call here:</font></div>
<div><font color="#500050" face="arial,
sans-serif"><br>
</font></div>
</div>
<div><font color="#500050" face="arial,
sans-serif">Capture from OpenSIPS:
xxxxxxxxxxxxxx</font></div>
<div><span style="color:rgb(80,0,80);font-family:arial,sans-serif">Capture
from Asterisk: xxxxxxxxxxxxxx</span></div>
<div>
<div>
<div><font color="#500050" face="arial, sans-serif"><br>
</font></div>
<div><font color="#500050" face="arial, sans-serif">Thanks,</font></div>
<div><font color="#500050" face="arial, sans-serif">Josh<br>
</font><br>
<div class="gmail_quote"> On Wed,
Mar 14, 2012 at 1:02 PM, Joshua
Nankin <span dir="ltr"><<a href="mailto:jnankin@gmail.com" target="_blank">jnankin@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"> Yeah, I'm
seeing that now. I just did a
capture and verified, the<br>
200s are coming out of my
Asterisk box, and OpenSIPS is
relaying them<br>
properly. I'm not getting the
ACK back from my carrier. I'm
going to<br>
check with them and send the
capture files I just generated -
they use<br>
IP whitelists, and this may just
be a authentication/security
problem.<br>
<br>
Thanks again!<br>
<br>
-Josh<br>
<br>
On Wed, Mar 14, 2012 at 12:59
PM, Bogdan-Andrei Iancu<br>
<div>
<div><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
wrote:<br>
> Hi Joshua,<br>
><br>
> It seams to be a lot of
200 OK retransmissions from
Asterisk because there<br>
> is no ACK from the
original caller side.<br>
><br>
> Do you see the caller
sending such ACK ? have you
tried to make a network<br>
> capture ?<br>
><br>
> Regards,<br>
> Bogdan<br>
><br>
><br>
><br>
> On 03/14/2012 06:27 PM,
Joshua Nankin wrote:<br>
>><br>
>> Hey Bogdan!<br>
>><br>
>> So, it appears that
the call gets to the LB and
is forwarded to<br>
>> Asterisk. Here's
what happens in both the LB
logs and Asterisk:<br>
>><br>
>> Asterisk - <a href="http://pastebin.com/" target="_blank">http://pastebin.com/</a><br>
>> OpenSIPS - <a href="http://pastebin.com/" target="_blank">http://pastebin.com/</a><br>
>><br>
>><br>
>> I have debug=6 on
OpenSIPS, so there's quite a
bit of logging<br>
>> happening for a
pretty short call. It seems
that there's a<br>
>> retransmission
problem here. Also, I'm
trying to negotiate T.38, so<br>
>> I'm not sure if
that has something to do
with this.<br>
>><br>
>> Thanks,<br>
>> Josh<br>
>><br>
>> On Wed, Mar 14,
2012 at 7:25 AM,
Bogdan-Andrei Iancu<br>
>> <<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
wrote:<br>
>>><br>
>>> Hi Joshua,<br>
>>><br>
>>> Have you
checked (at network level)
if the actual call (the
inbound one,<br>
>>> from a client)
gets to opensips LB ? and if
so, if the call gets<br>
>>> forwarded<br>
>>> to Asterisk ?<br>
>>><br>
>>> Regards,<br>
>>> Bogdan<br>
>>><br>
>>><br>
>>> On 03/13/2012
10:00 PM, Joshua Nankin
wrote:<br>
>>>><br>
>>>> Hello,<br>
>>>><br>
>>>> Trying to
use OpenSIPS as both a load
balancer for incoming calls
and<br>
>>>> as an
outbound proxy to my
carrier. I have been
unsuccessful to place<br>
>>>> outgoing or
incoming calls, and I think
I just need some help with
the<br>
>>>>
configuration. Just for
testing, I have one Asterisk
box behind<br>
>>>> OpenSIPS
and the load balancer table
on OpenSIPS looks like this:<br>
>>>><br>
>>>><br>
>>>><br>
>>>>
+----+----------+-------------------+-----------+------------+-------------+<br>
>>>> | id |
group_id | dst_uri
| resources | probe_mode |<br>
>>>> description<br>
>>>> |<br>
>>>><br>
>>>><br>
>>>>
+----+----------+-------------------+-----------+------------+-------------+<br>
>>>> | 1 |
1 | <a>sip:10.36.115.119</a>
| fax=300 | 0 |
tester<br>
>>>> |<br>
>>>><br>
>>>><br>
>>>>
+----+----------+-------------------+-----------+------------+-------------+<br>
>>>><br>
>>>><br>
>>>> The dst_ur
is the internal IP of my
asterisk box. I'm using
this for<br>
>>>> fax, so I
have set the resources field
to 300 channels of "fax".
On<br>
>>>> my asterisk
box, I've added
"outboundproxy=23.21.170.154",
as the 23.*<br>
>>>> IP is the
external IP of the opensips
box. Below is my simple<br>
>>>>
opensips.cfg that I've
adapted from the load
balancing tutorial. Any<br>
>>>> help would
be much appreciated.
Additionally, for debugging
purposes,<br>
>>>> I've opened
all UDP and TCP ports.
Still not able to connect.<br>
>>>><br>
>>>><br>
>>>> debug=6<br>
>>>> memlog=1<br>
>>>><br>
>>>> fork=yes<br>
>>>> children=2<br>
>>>>
log_stderror=yes<br>
>>>>
log_facility=LOG_LOCAL0<br>
>>>><br>
>>>>
disable_tcp=yes<br>
>>>>
disable_dns_blacklist = yes<br>
>>>><br>
>>>>
auto_aliases=no<br>
>>>><br>
>>>>
check_via=no<br>
>>>> dns=off<br>
>>>> rev_dns=off<br>
>>>> port=5060<br>
>>>><br>
>>>>
mpath="/usr/lib/opensips/modules/"<br>
>>>> loadmodule
"maxfwd.so"<br>
>>>> loadmodule
"sl.so"<br>
>>>> loadmodule
"db_mysql.so"<br>
>>>> loadmodule
"tm.so"<br>
>>>> loadmodule
"uri.so"<br>
>>>> loadmodule
"rr.so"<br>
>>>> loadmodule
"dialog.so"<br>
>>>> loadmodule
"mi_fifo.so"<br>
>>>> loadmodule
"signaling.so"<br>
>>>> loadmodule
"textops.so"<br>
>>>> loadmodule
"load_balancer.so"<br>
>>>><br>
>>>>
modparam("mi_fifo",
"fifo_name",
"/tmp/opensips_fifo")<br>
>>>>
modparam("mi_fifo",
"fifo_mode",0666)<br>
>>>><br>
>>>>
modparam("dialog",
"db_mode", 1)<br>
>>>>
modparam("dialog", "db_url",
"mysql://root:opensips@localhost/opensips")<br>
>>>><br>
>>>>
modparam("rr","enable_double_rr",1)<br>
>>>>
modparam("rr","append_fromtag",1)<br>
>>>><br>
>>>>
modparam("load_balancer",<br>
>>>>
"db_url","mysql://root:opensips@localhost/opensips")<br>
>>>><br>
>>>><br>
>>>> route{<br>
>>>> if
(!mf_process_maxfwd_header("3"))
{<br>
>>>>
sl_send_reply("483","looping");<br>
>>>>
exit;<br>
>>>> }<br>
>>>><br>
>>>><br>
>>>> if
(!has_totag()) {<br>
>>>>
# initial request<br>
>>>>
record_route();<br>
>>>> }
else {<br>
>>>>
# sequential request
-> obey Route
indication<br>
>>>>
loose_route();<br>
>>>>
t_relay();<br>
>>>>
exit;<br>
>>>> }<br>
>>>><br>
>>>> #
handle cancel and
re-transmissions<br>
>>>> if (
is_method("CANCEL") ) {<br>
>>>>
if ( t_check_trans() )<br>
>>>>
t_relay();<br>
>>>>
exit;<br>
>>>> }<br>
>>>><br>
>>>><br>
>>>> #
from now on we have only the
initial requests<br>
>>>> if
(!is_method("INVITE")) {<br>
>>>>
send_reply("405","Method
Not Allowed");<br>
>>>>
exit;<br>
>>>> }<br>
>>>><br>
>>>> #
detect resources and do
balancing<br>
>>>>
load_balance("1","fax");<br>
>>>><br>
>>>> #
LB function returns negative
if no suitable destination
(for<br>
>>>> requested
resources) is found,<br>
>>>> #
or if all destinations are
full<br>
>>>> if
($retcode<0) {<br>
>>>>
sl_send_reply("500","Service
full");<br>
>>>>
exit;<br>
>>>> }<br>
>>>><br>
>>>>
xlog("Selected destination
is: $du\n");<br>
>>>><br>
>>>> #
send it out<br>
>>>> if
(!t_relay()) {<br>
>>>>
sl_reply_error();<br>
>>>> }<br>
>>>> }<br>
>>>><br>
</div>
</div>
</blockquote>
</div>
</div>
</div>
</div>
</blockquote>
<br>
<div>
<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>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<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>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<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></div>
</blockquote></div><br></div>