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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Perhaps showing ignorance here, but this
sounds like something call-limit in sip.conf could fix. You have 5 agents all
on SIP extensions answering queues from incoming SIP trunks. I’ve
experienced a situation where Zap trunks would let you talk on Asterisk and a
Legacy PBX simultaneously, but it would seem that SIP trunks on the same switch
would be aware of one another.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>David @ULC<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, April 16, 2009
11:37 AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users]
Simultaneous Calls at a time</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Even I thought so thats
why I tried with 4 VOIP provider and things didn't change. :-(<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>On Thu, Apr 16, 2009 at 8:36 PM, David @ULC <<a
href="mailto:ucoms2001@gmail.com">ucoms2001@gmail.com</a>> wrote:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>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 border=0 width=15 height=15 id="_x0000_i1025"
src="http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3dpbmsuZ2lm&b=2"
alt=Wink> } </span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=1 face=Verdana><span style='font-size:9.0pt;
font-family:Verdana'>Also, this thing happens even when we have just 5 agents
on a single server. <img border=0 width=15 height=15 id="_x0000_i1026"
src="http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vaW1hZ2VzL3NtaWxlcy9pY29uX3NhZC5naWY%3D&b=2"
alt=Sad> <br>
<br>
Our version is Asterisk 1.2.27</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=1 face=Verdana><span style='font-size:9.0pt;
font-family:Verdana'><br>
Any Solutions ?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
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