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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><FONT
face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </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><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=822340623-28012008><FONT face=Arial size=2>Any suggestions are
very welcome.</FONT></SPAN></DIV>
<DIV><SPAN class=822340623-28012008><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=822340623-28012008><FONT face=Arial
size=2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=822340623-28012008><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=822340623-28012008><FONT face=Arial
size=2>Daniel</FONT></SPAN></DIV></BODY></HTML>