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<tt>Hi Mikhail,<br>
<br>
1) I'm aware of the issue with the silence during the agent
ringing - to be honest, we did some scripting workaround to use
rtpproxy to inject a ringback tone to caller while we ring the
agent :D.<br>
But the proposed solution is definitely better, still it needs to
be evaluated if possible or not from the coding perspective (more
or less, it depends on how the b2b works in OpenSIPS) <br>
Could you please take this report/issue/topic and create a new
issue/ticket on the github tracker, so we can have it on the
roadmap ?<br>
<br>
2) I suppose the fault is related to frame 35-37 in
callcenter.pcap, right ? Also please open a ticket here and attach
the pcap + opensips logs in level 4 (dbg).<br>
<br>
Best regards,<br>
</tt>
<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>
OpenSIPS Summit 2019
<a class="moz-txt-link-freetext" href="https://www.opensips.org/events/Summit-2019Amsterdam/">https://www.opensips.org/events/Summit-2019Amsterdam/</a>
</pre>
<div class="moz-cite-prefix">On 05/23/2019 08:25 PM, Mikhail wrote:<br>
</div>
<blockquote type="cite"
cite="mid:55da4fab-ed98-5711-b49c-b6a7267250dd@yandex.ru">
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Hi Bogdan-Andrei,<br>
<br>
<p>1)</p>
Right. I'll try to explain more detailed what I mean:<br>
<br>
case 1 - as I see how opensips working<br>
<br>
Caller calls and Agent is not available<br>
caller is listening intro<br>
caller is listening music<br>
Agent become available<br>
Music stops, caller listening silence - Agent ringing<br>
Agent don't answer within within defined period, for example 20
seconds, and Caller is listening silence 20 seconds!<br>
Call returned to queue<br>
Caller starts to listening music from beginning, because opensips
makes new call to ivr server<br>
Agent become available<br>
Music stops, caller is listening silence - Agent ringing<br>
in a 10 seconds Agent answers, so 10 seconds caller listen silence
again.<br>
<br>
<br>
case 2 - as I know how it should be<br>
<br>
Caller calls and Agent is not available<br>
caller is listening intro<br>
caller is listening music<br>
Agent become available<br>
Music don't stops, caller continue listening music without
interrupt - Agent ringing<br>
Agent don't answer within within defined period, for example 20
seconds, but Caller do not know about it, he sill listening music
without interrupt.<br>
Call returned to queue, caller continue listening music without
interrupt.<br>
Agent become available<br>
Music don't stops, caller continue listening music without
interrupt - Agent ringing<br>
in a 10 seconds Agent answers, at this point caller stop listening
music and start talking to agent.<br>
<p><br>
</p>
<p>2)<br>
</p>
<p>I made 2 calls for test:<br>
1-st - direct call from caller to agent. Hold by caller and
agent works good.<br>
2-nd - call from caller to number 3001 routed to ivr by
cc_handle_call() command<br>
call routed to intro 0081 and then to agent. Hold did not work.<br>
I found that it seems that agent’s xLite did no receive some sip
reply and after 30 seconds after answer xLite sends bye by
itself. <br>
Also there are errors in opensips.log<br>
</p>
<p>sip log captured with sngrep I uploaded to my server:<br>
</p>
<p>video2dv.com/download/opensips/direct.pcap<br>
video2dv.com/download/opensips/direct.txt<br>
video2dv.com/download/opensips/callcenter.pcap<br>
video2dv.com/download/opensips/callcenter.txt<br>
video2dv.com/download/opensips/callcenter.log<br>
<br>
</p>
<pre class="moz-signature" cols="72">Regards,
Laba Mikhail
</pre>
<div class="moz-cite-prefix">22.05.2019 10:09, Bogdan-Andrei Iancu
wrote:<br>
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<blockquote type="cite"
cite="mid:7c5c9dbd-71a1-4e52-4262-8642e901ae72@opensips.org">
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<tt>Hi Anton,<br>
<br>
In regards to your experience with the call_center module:<br>
<br>
1) You mean, while the call is ringing on the agent side, the
caller gets dead-air, right ?<br>
<br>
2) It should be working, do you have a pcap on that ?<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="https://www.opensips-solutions.com" moz-do-not-send="true">https://www.opensips-solutions.com</a>
OpenSIPS Summit 2019
<a class="moz-txt-link-freetext" href="https://www.opensips.org/events/Summit-2019Amsterdam/" moz-do-not-send="true">https://www.opensips.org/events/Summit-2019Amsterdam/</a>
</pre>
<div class="moz-cite-prefix">On 04/28/2019 02:34 PM, Mikhail
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:bb9e62e8-b3f9-b233-184c-3fcc7e7a5032@yandex.ru">
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<p>Hi Антон,</p>
<p>in my tests I have working greeting and queue and I sill
have issues I described before:</p>
<p>- when Invite is going to agent, caller hear silence,
because opensips stops call to medial server. <br>
</p>
<p>- can't place on Hold either agent or caller, opensips
sends "400 Not Acceptable".</p>
<p>I guess this is a feature/bugs in call_center module.</p>
<p>My goal is to manage call_center with more then 1000 agents
and migrate from our current platform, but this troubles
stops moving.<br>
</p>
<p><span class="tlid-translation translation" lang="en"><span
title="" class="">It would be very cool if the
developers or those who use the </span></span><span
class="tlid-translation translation" lang="en"><span
title="" class="">call_center module have commented on
these problems.</span></span></p>
<p><span class="tlid-translation translation" lang="en"><span
title="" class=""><br>
</span></span></p>
<p>Laba Mikhail<br>
</p>
<p><br>
</p>
25.04.2019 16:12, Антон Ершов пишет:<br>
<blockquote type="cite"
cite="mid:CAMp8X_vc+gx5h3c_8VTpYcoxM9tgnoj_vS7a=rUb4Vqz=JSf5A@mail.gmail.com">
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<pre class="m_-3630660128883277857gmail-tw-data-text m_-3630660128883277857gmail-tw-ta m_-3630660128883277857gmail-tw-text-medium" id="m_-3630660128883277857gmail-tw-target-text" dir="ltr" style="unicode-bidi:isolate;border:none;padding:0px 0.14em 0px 0px;margin-top:0px;margin-bottom:0px;font-family:inherit;overflow:hidden;width:275px;white-space:pre-wrap;color:rgb(33,33,33);height:108px;font-size:29px;line-height:36px"><pre class="m_-3630660128883277857gmail-tw-data-text m_-3630660128883277857gmail-tw-ta m_-3630660128883277857gmail-tw-text-medium" id="m_-3630660128883277857gmail-tw-target-text" dir="ltr" style="unicode-bidi:isolate;border:none;padding:0px 0.14em 0px 0px;margin-top:0px;margin-bottom:0px;font-family:inherit;overflow:hidden;width:275px;white-space:pre-wrap;height:108px;line-height:36px"><pre class="gmail-tw-data-text gmail-tw-ta gmail-tw-text-medium" id="gmail-tw-target-text" dir="ltr" style="unicode-bidi:isolate;border:none;padding:0px 0.14em 0px 0px;margin-top:0px;margin-bottom:0px;font-family:inherit;overflow:hidden;width:275px;white-space:pre-wrap;height:108px;line-height:36px"><span tabindex="0" lang="en">hi, did you manage to launch the call center module?</span></pre></pre></pre>
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