I see also something like this:<br><br>Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4547]: sending call with callid <a href="mailto:3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10">3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10</a> and ruri <a href="mailto:sip%3A0758553307@192.168.52.20">sip:0758553307@192.168.52.20</a> to sip:192.168.254.241<br>
Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4539]: sending call with callid <a href="mailto:3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10">3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10</a> and ruri <a href="mailto:sip%3A0758553307@192.168.52.20">sip:0758553307@192.168.52.20</a> to sip:192.168.254.241<br>
Mar 15 18:47:36 opensipsvl /usr/local/sbin/opensips[4556]: sending call with callid <a href="mailto:3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10">3b77e3325a8c75f8571c4fcf37a8ba3c@192.168.52.10</a> and ruri <a href="mailto:sip%3A0758553307@192.168.52.20">sip:0758553307@192.168.52.20</a> to sip:192.168.254.242<br>
<br>Same call 3 times.. 2 to one server and another time to the other<br><br><div class="gmail_quote">On Tue, Mar 15, 2011 at 12:10 PM, 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi Iulian,<br>
<br>
<br>
<br>
Related to the second issue (with 2 logs for a single call), I suspect it may be a retransmission issue. To be sure, print in your xlog the callid also (to be sure it is the same request) . Var for callid is $ci.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
<br>
Iulian Macare 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 class="im">
Hello<br>
<br>
<br>
I have installed OpenSips 1.6.4 on CentOS 5.5 32bit with load balancing & mysql support ; I want to balance calls to 2 asterisk servers . I am sending traffic to opensips from 1 x gnudialer & 1 x vicidial ( so from predictive dialers ). Situation is like this:<br>
<br>
<br>
+----+----------+--------------------------+-----------+------------+-------------+<br>
| id | group_id | dst_uri | resources | probe_mode | description |<br>
+----+----------+--------------------------+-----------+------------+-------------+<br></div>
| 1 | 1 | sip:<a href="http://192.168.254.241:5060" target="_blank">192.168.254.241:5060</a> <<a href="http://192.168.254.241:5060" target="_blank">http://192.168.254.241:5060</a>> | pstn=300 | 0 | |<br>
| 2 | 1 | sip:<a href="http://192.168.254.242:5060" target="_blank">192.168.254.242:5060</a> <<a href="http://192.168.254.242:5060" target="_blank">http://192.168.254.242:5060</a>> | pstn=300 | 0 | |<div>
<div></div><div class="h5"><br>
+----+----------+--------------------------+-----------+------------+-------------+<br>
<br>
600 channels in total , and I send around 500 channels ; OpenSips drops a lot of calls; By drop I mean a call that is not sent to one of those 2 asterisk servers that I have.<br>
<br>
The code for balancing in this situation is:<br>
<br>
if (uri=~"^sip:0[1-9][0-9]*@") {<br>
load_balance("1","pstn");<br>
xlog("sending call $ru to $du\n");<br>
t_relay();<br>
exit;<br>
}<br>
<br>
! An important thing to say is that in /var/log/messages I see the specific number that is sent to 192.168.254.241 for example ; So the parameter xlog("sending call $ru to $du\n"); works ; The problem is that in logs on 192.168.254.241 that number never arrives in asterisk logs ; In logs of vicidial & gnudialer I see it like congestion.<br>
<br>
If I do something like this:<br>
<br>
+----+----------+---------------------+-----------+------------+-------------+<br>
| id | group_id | dst_uri | resources | probe_mode | description |<br>
+----+----------+---------------------+-----------+------------+-------------+<br>
| 1 | 1 | sip:192.168.254.241 | pstn=150 | 0 | |<br>
| 2 | 2 | sip:192.168.254.241 | pstn=150 | 0 | |<br>
| 3 | 1 | sip:192.168.254.242 | pstn=150 | 0 | |<br>
| 4 | 2 | sip:192.168.254.242 | pstn=150 | 0 | |<br>
<br>
And I split opensips balancing in 2<br>
<br>
<br>
if(src_ip==192.168.3.10 )<br>
{<br>
load_balance("1","pstn");<br>
xlog("sending call to $du\n");<br>
t_relay();<br>
exit;<br>
};<br>
<br>
if(src_ip==192.168.3.11 )<br>
{<br>
load_balance("2","pstn");<br>
xlog("sending call to $du\n");<br>
t_relay();<br>
exit;<br>
};<br>
<br>
<br>
and by doing this I get the same numbers of channels on opensips ( around 500 channels ) but I am splitting in 2 groups of load balancing; It can process all the calls.<br>
<br>
Another question that I saw is that when I make a single call to opensips and I involve load balancing in /var/log/messages I get 2 times the same message .. just like it send 2 time to asterisk server the call .. but on asterisk I receive only one time.<br>
<br>
Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to sip:192.168.254.241<br>
Mar 10 14:58:47 opensips /usr/local/sbin/opensips[27611]: sending call to sip:192.168.254.241<br>
<br>
<br>
Isn't load balancing fast enough the process the calls made by predictive dialers, when over 300 channels is sent .. ? Or I have some mistakes made .<br>
<br>
<br></div></div>
------------------------------------------------------------------------<br>
<br>
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<br>
</blockquote>
<br>
<br>
-- <br>
Bogdan-Andrei Iancu<br>
OpenSIPS eBootcamp - 28th February 2011<br>
OpenSIPS solutions and "know-how"<br>
<br>
<br>
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