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<tt>Hi Dragomir,<br>
<br>
There is a difference between a Call Center service and the Call
Center module - the Call Center module in OpenSIPs is a call
queuing with distribution. <br>
<br>
And I didn't mention adding more servers, I said to use 2
different opensips instances (on same server, different port),
just for a better script separation and to simplify your work. Of
course, you can still do it with a single instance, as Razvan
suggested, if you have some good scripting skills.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
</pre>
<div class="moz-cite-prefix">On 11/16/2017 04:57 PM, Dragomir
Haralambiev wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CADzurOKjpoY11sFRbzPgE+DXMc7waEfJMFNOPOCNgYpr_eMuMw@mail.gmail.com">
<div dir="ltr">
<div>Hi Bogdan,</div>
<div><br>
</div>
<div>Thanks for your replay.</div>
<div><br>
</div>
<div>When speek about callcenter service I think the basic
functionality must be: outgoing calls, incoming calls, voice
recording and transcoding (change the voice codec).</div>
<div><br>
</div>
<div>How to solve this case?</div>
<div>We have the option to choose one of the following two
solutions.</div>
<div><br>
</div>
<div>1. Increase the number of servers (what you offer).</div>
<div><br>
</div>
<div>To do this I must to add other one server(service) for
recording and transcoding like Asterisk or Freeswitch.</div>
<div><br>
</div>
<div>Opensips_1 <-----> Opensips_2 <------> Asterisk
or FreeSwitch</div>
<div><br>
</div>
<div>Total Efficiency =
Efficiency(Opensips_1).Efficiency(Opensips_2).Efficiency(Asterisk)=0,9
. 0,9 . 0,9 =0.72 (72%).</div>
<div><br>
</div>
<div>This shows theoretically that a horizontal increase in
servers will result in a loss of 25% working capacity.</div>
<div><br>
</div>
<div>2. To use one server (service) to solve this case.</div>
<div><br>
</div>
<div>Until now I have been using Opensipts, but he can not solve
these problems.</div>
<div>Unfortunately I have to use Asterisk or Freeswich for
callcenter service.</div>
<div><br>
</div>
<div>Best regards,</div>
<div>Dragomir</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2017-11-16 14:10 GMT+02:00
Bogdan-Andrei Iancu <span dir="ltr"><<a
href="mailto:bogdan@opensips.org" target="_blank"
moz-do-not-send="true">bogdan@opensips.org</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Dragomir,<br>
<br>
The Call Center module handles the agents as simple SIP
URIs. if in your case the agents are actually
registering devices, this is a different matter. What I
suggest, for simplicity reasons is to use 2 opensips
instances. <br>
<br>
The first to be a dedicated call center seeing the
agents as AoRs (<a
class="m_-4745286077269750118moz-txt-link-freetext"
moz-do-not-send="true">sip:user@domain</a>) where the
domain points to the IP of the second server<br>
<br>
The second instance to be simple registrar server for
the "domain" where the agents are registering their
devices.<br>
<br>
In terms of call you will have:<br>
<br>
---(call to queue)----> OPENSIPS1 -----(call to agent
user@domain)----> OPENSIPS2 -----(call to registered
device)-----> AGENT DEVICE<br>
<br>
<br>
Regards,<br>
</tt>
<pre class="m_-4745286077269750118moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-4745286077269750118moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank" moz-do-not-send="true">http://www.opensips-solutions.<wbr>com</a>
</pre>
<div>
<div class="h5">
<div class="m_-4745286077269750118moz-cite-prefix">On
11/10/2017 04:21 PM, Dragomir Haralambiev wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>Hi,</div>
<div><br>
</div>
<div>I try your idea to loop the INVITE back to
Opensips and do a lookup(). I pit into
"location" field from cc_agents table this one <a
class="m_-4745286077269750118moz-txt-link-rfc2396E"
moz-do-not-send="true">"sip:localnumber@Opensips_IP:<wbr>Opensips_port"</a>.</div>
<div><br>
</div>
<div>Call is connected but I receive follow log
message:</div>
<div><br>
</div>
<div> ERROR:b2b_entities:b2b_<wbr>prescript_f: No
dialog found, callid=
[B2B.337.5360955.1510320840], method=OPTIONS</div>
<div><br>
</div>
<div>When call is finished record from dialog
tables is not deleted. <br>
<br>
Where is problem?</div>
<div><br>
</div>
<div>Regards,</div>
<div>Dragomir</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2017-11-06 11:37
GMT+02:00 Răzvan Crainea <span dir="ltr"><<a
href="mailto:razvan@opensips.org"
target="_blank" moz-do-not-send="true">razvan@opensips.org</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <tt>Hi,
Hristo!<br>
<br>
1. You are right, b2b_entities should be
loaded as a dependency for b2b_logic. You
may argue that it should also be listed as
a dependency to call_center as well, but
indirectly.<br>
<br>
2. Yes, you should populate the tables
with the flows you want to support and the
agents that are going to use the call
center.<br>
<br>
3. There is currently no way to directly
set the location of the agent dynamically.
However, we can do some work-arounds, for
example always advertise yourself as the
location of the user, and for each call
you can loop the INVITE back to you and do
a lookup(). Or simply use an external
registrar service.<br>
<br>
Best regards,<br>
</tt>
<pre class="m_-4745286077269750118m_8082184747567676002moz-signature" cols="72">Răzvan Crainea
OpenSIPS Developer
<a class="m_-4745286077269750118m_8082184747567676002moz-txt-link-abbreviated" href="http://www.opensips-solutions.com" target="_blank" moz-do-not-send="true">www.opensips-solutions.com</a></pre>
<div>
<div class="m_-4745286077269750118h5">
<div
class="m_-4745286077269750118m_8082184747567676002moz-cite-prefix">On
11/05/2017 11:06 PM, Hristo Donev
wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div class="m_-4745286077269750118h5">
<div dir="ltr">
<div>Hello All,</div>
<div><br>
</div>
<div>I try to implement call_center
module in Opensips 2.3.</div>
<div>Here are my questions?</div>
<div><br>
</div>
<div>1. From the documentation
1.3.1:</div>
<div>"b2b_logic" and "database" must
be loaded before callcenter
module.</div>
<div><br>
</div>
<div>I found out that "b2b_entities"
should be loaded as well,
otherwise the opensips does not
start.</div>
<div><br>
</div>
<div>Is this my mistake or a
documentation error?</div>
<div><br>
</div>
<div>2. Opensips can not start if
the tables "cc_agents" and
"cc_flow" are empty.</div>
<div>Why it cannot start? Should
they be populated with some data?</div>
<div><br>
</div>
<div>3. Table "cc_agents", field
"location" should point to the SIP
agent location.</div>
<div>If the agent's IP and port are
changed dynamically -</div>
<div>how to copy automatically
"contact" field from "location"
table to "cc_agents" field
"location"? </div>
<div><br>
</div>
<div>I would be grateful to receive
a response for my questions.</div>
<div><br>
</div>
<div>Thanks in advance,</div>
<div>Hristo Donev</div>
</div>
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