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Why not give the receptionist a two line phone? Register one line on
server 1 and the other on server 2. Then the bounce back and forth
goes away saving bandwidth.<br>
<br>
Lyle<br>
<br>
Daniel Cole wrote:
<blockquote
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type="cite">
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<div><font face="Arial" size="2"><span class="822340623-28012008">Hello
List,</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">I
am currently having a bit of a strange issue with a pair of asterisk
servers that we recently set up.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">For
a bit of background, this particular business has two sites in two
different towns, about 10 minutes apart. They have 3 analogue PSTN
lines connected to the asterisk servers at each location, via a Sangoma
A200 (with HEC). They are trying to have just the one receptionist for
the whole organization, answering calls that come in for both locations.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">We
have a problem where some calls (seemingly randomly) appear to get one
way audio. This only happens for inbound calls off the PSTN, if they
follow this pattern (which is a fair number of calls):</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">Call
comes in from PSTN to site A, gets put into a queue to be answered by
receptionist as site B. Receptionist answers the call, and then puts
the call on hold to perform an attended transfer to an extension at
site A. (The call from the receptionist to the extension is OK). When
the receptionist hits the 'transfer' button to actually transfer the
call, the original caller cannot hear anything. The internal extension
can hear the caller OK.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">This
problem does not occur on every call. Since the issue has risen its
head, I have enabled core, sip and iax debugging, but I am of yet
unable to get the issue to occur on its own, to have a good look at the
log files.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">FYI,
I have disabled the asterisk Dial Commands in FreePBX, to solve another
issue (where call audio bounces between the servers for a call that is
transferred between sites and back again).</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">Both
servers are asterisk version 1.2.23, freepbx version 2.3.1.0.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008">I
have posted the contents of the iax.conf file below (which is identical
on both servers). If there is any further information I can provide,
please let me know and I can get this information.</span></font></div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div><font face="Arial" size="2"><span class="822340623-28012008"></span></font> </div>
<div> </div>
<div><font face="Arial" size="2">[general]</font></div>
<div> </div>
<div><font face="Arial" size="2">disallow=all<br>
allow=g729<br>
mailboxdetail=yes</font></div>
<div> </div>
<div><font face="Arial" size="2">jitterbuffer=no<br>
;maxjitterbuffer=500<br>
;jittershrinkrate=1<br>
bandwidth=low<br>
tos=lowdelay<br>
trunk=yes<br>
notransfer=yes</font></div>
<div> </div>
<div><font face="Arial" size="2">#include iax_general_custom.conf<br>
#include iax_registrations_custom.conf<br>
#include iax_registrations.conf<br>
#include iax_custom.conf<br>
#include iax_additional.conf<br>
</font></div>
<div> </div>
<div> </div>
<div> </div>
<div><span class="822340623-28012008"><font face="Arial" size="2">Any
suggestions are very welcome.</font></span></div>
<div><span class="822340623-28012008"></span> </div>
<div><span class="822340623-28012008"><font face="Arial" size="2">Regards,</font></span></div>
<div><span class="822340623-28012008"></span> </div>
<div><span class="822340623-28012008"><font face="Arial" size="2">Daniel</font></span></div>
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