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<tt>Hi Nabeel,<br>
<br>
OpenSIPS does not assume anything by default. If you want to have
any new calls to user A rejected (if A already in a call, with
other users or any service), you need to script this.<br>
You should use dialog profiles to count the ongoing calls for the
user A and when you get a new call for A and the profile counter
is <> 0, just do the reject.<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="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 08.08.2016 01:44, Nabeel wrote:<br>
</div>
<blockquote
cite="mid:CA+vx6KKU+=zf9-jVji_UF2VUsrnuA3R-ZsZxuOVpvrm8Q2UN8A@mail.gmail.com"
type="cite">
<p dir="ltr">Hi,</p>
<p dir="ltr">The busy reply should be generated in exactly the
same way as when the callee is busy on the phone with another
SIP user (from the OpenSIPS subscriber table). The only
difference is that the callee is on the phone with asterisk
voicemail instead of a SIP user from the subscriber table. I'm
not sure who exactly should generate the reply but I think it
depends on how OpenSIPS is interacting with the Voicemail server
during a call to voicemail. Perhaps it's not treating the
voicemail service as another 'user' so the caller didn't receive
the busy signal.</p>
<p dir="ltr">Nabeel</p>
<div class="gmail_extra"><br>
<div class="gmail_quote">On 7 Aug 2016 6:57 p.m., "Bogdan-Andrei
Iancu" <<a moz-do-not-send="true"
href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>>
wrote:<br type="attribution">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Hi Nabeel,<br>
<br>
Who should generate the 486 Busy reply ?<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a></pre>
<div>On 06.08.2016 02:56, Nabeel wrote:
</div>
<blockquote type="cite">
<div dir="ltr">Hi,
<div>
</div>
<div>OpenSIPS does not receive a '486 Busy' signal when a callee
is using the Asterisk voicemail service. If a user is
currently listening to voicemail or leaving a voice message,
an attempt to call that user should result in a '486 Busy'
signal. From that response, I will be able to play a busy tone
on the caller's side. Please let me know how to fix this.</div>
<div>
</div>
<div>Nabeel</div>
</div>
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</pre>
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