<span class="Apple-style-span" style="font-family: 'Times New Roman'; font-size: 16px; "><pre>busy-level ?</pre><pre>How to use it and whats the purpose ?</pre></span><br><div class="gmail_quote">On Thu, Apr 16, 2009 at 10:43 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><br></div><a href="http://threebit.net/mail-archive/asterisk-users/msg07138.html" target="_blank">http://threebit.net/mail-archive/asterisk-users/msg07138.html</a><div>
<br></div><div><span style="font-family:'Courier New';font-size:12px">Remember that if you want to support attended transfers, you need at least two<br>
simultaneous calls.</span></div><div><span style="font-family:'Courier New';font-size:12px"><br></span></div><div><span style="font-family:'Courier New';font-size:12px">So, its safe bet to keep call-limit=2.</span></div>
<div><span style="font-family:'Courier New';font-size:12px"><br></span>Advice ?</div><div><div></div><div class="h5"><div><br></div><div><br><div class="gmail_quote">On Thu, Apr 16, 2009 at 10:37 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>My SIP config is below :</div><div><br></div><div>[sip64]</div><div>type=peer</div><div>username=fiduci</div><div>
fromuser=fiduci</div><div>authuser=fiduci</div><div>secret=pass</div><div>host=64.33.22.11</div><div>nat=no</div>
<div>canreinvite=yes</div><div>insecure=very</div><div>disallow=all</div><div>allow=g729</div><div>allow=ulaw</div><div>context=default</div><div>dtmfmode=rfc2833</div><div><br></div><div>Now, I need to add another element as <span style="font-family:-webkit-monospace;font-size:16px;white-space:pre">call-limit=1 and this should solve my problem ?</span><br>
</div><div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre"><br></span></div><div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre">If yes. Great. Kindly advice.</span></div>
<div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre"><br></span></div><div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre">But will that allow 3 party conference ?</span></div>
<div><div></div><div>
<div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre"><br></span></div><div><span style="font-family:-webkit-monospace;font-size:16px;white-space:pre"><br>
</span><div class="gmail_quote">On Thu, Apr 16, 2009 at 10:22 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span style="font-family:'Times New Roman';font-size:16px"><pre>"call-limit in sip.conf"</pre><pre>Can you elaborate please and how to set that.</pre><pre>Lets presume I have 10 agents and dial ratio is 4.</pre>
</span><div><div></div><div><br><div class="gmail_quote">On Thu, Apr 16, 2009 at 10:06 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><br></div>Even I thought so thats why I tried with 4 VOIP provider and things didn't change. :-(<div><div></div><div><br><br><div class="gmail_quote">On Thu, Apr 16, 2009 at 8:36 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
<div><span style="font-family:Verdana;font-size:16px"><br></span></div><div><span style="font-family:Verdana;font-size:16px"><span style="font-size:12px;line-height:18px">Many time we face an issue where even if an agent is on Call, another call comes in. <br>
<br>Sometimes, even if agent hang up the call, call stays back and another come sin and then both customers can hear each other { which i think is VERY dangerous <img src="http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3dpbmsuZ2lm&b=2" alt="Wink" border="0"> } <br>
<br></span></span></div><div><span style="font-family:Verdana;font-size:16px"><span style="font-size:12px;line-height:18px">Also, this thing happens even when we have just 5 agents on a single server. <img src="http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3NhZC5naWY%3D&b=2" alt="Sad" border="0"> <br>
<br>Our version is Asterisk 1.2.27</span></span></div><div><span style="font-family:Verdana;font-size:16px"><span style="font-size:12px;line-height:18px"><br>
Any Solutions ?</span><br></span></div><div><span style="font-family:Verdana;font-size:12px;line-height:18px"><br></span></div><div><span style="font-family:Verdana;font-size:12px;line-height:18px"><br>
</span></div>
</blockquote></div><br>
</div></div></blockquote></div><br>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br></div>
</div></div></blockquote></div><br>