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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'>Ok, I finally got asterisk to output the &#8216;full&#8217;
log so I could review it and it finally dropped a call on me. As you can see in
the log below, asterisk thinks it is seeing the busy signal during a call that
has been in progress for over 30 seconds, I have set busydetect=no in my Zapata.conf
file, but wonder if anyone else agrees with this. This is the output I see when
it dropped the call:<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>&nbsp;</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'>2004-12-08 15:19:19 DEBUG[4650]: Device
'SIP/136' changed to state '2'<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'>2004-12-08 15:19:23 DEBUG[4650]: Acked
pending invite 102<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'>2004-12-08 15:19:23 DEBUG[4650]: Stopping
retransmission on '0ab2e6632b48cdaf7479ce5d7d277c30@192.168.4.25' of Request
102: Found<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'>2004-12-08 15:19:23 DEBUG[4650]: Oooh, we
need to change our formats since our peer supports only 0x4(ULAW) and not
0x2(GSM)<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'>2004-12-08 15:19:23 DEBUG[4650]:
build_route: Contact hop: &lt;sip:136@192.168.4.85&gt;<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'>2004-12-08 15:19:23
VERBOSE[4650]:&nbsp;&nbsp;&nbsp;&nbsp; -- SIP/136-8aa2 answered Zap/1-1<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'>2004-12-08 15:19:23 DEBUG[4650]: Requested
indication -1 on channel Zap/1-1<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'>2004-12-08 15:19:23 DEBUG[4650]: Ooh,
format changed from UNKN to ULAW<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'>2004-12-08 15:20:53 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:07 DEBUG[4650]: Device
'SIP/135' changed to state '0'<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'>2004-12-08 15:21:18 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:18 DEBUG[4650]: Stopping
retransmission on '66d2b161133ebf6810f91ec16e5081da@192.168.4.25' of Request
102: Found<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'>2004-12-08 15:21:22 DEBUG[4650]: Auto
destroying call '3812ff91-7ea5ee5b-8188aafe@192.168.4.86'<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'>2004-12-08 15:21:30 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:30 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:36 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:36 DEBUG[4650]: Setting
NAT on RTP to 0<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'>2004-12-08 15:21:50 DEBUG[4650]:
Requesting Hangup because the busy tone was detected on channel Zap/1-1<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'>2004-12-08 15:21:50 DEBUG[4650]: Didn't
get a frame from channel: Zap/1-1<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'>2004-12-08 15:21:50 DEBUG[4650]: Bridge
stops bridging channels Zap/1-1 and SIP/136-8aa2<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'>2004-12-08 15:21:50 DEBUG[4650]:
update_user_counter(136) - decrement outUse counter<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'>2004-12-08 15:21:50 DEBUG[4650]: Exiting
with DIALSTATUS=ANSWER.<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'>2004-12-08 15:21:50 VERBOSE[4650]:&nbsp;&nbsp;
== Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Zap/1-1' in macro
'stdexten'<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'>2004-12-08 15:21:50
VERBOSE[4650]:&nbsp;&nbsp; == Spawn extension (office-day, 136, 6) exited
non-zero on 'Zap/1-1'<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'>2004-12-08 15:21:50 DEBUG[4650]: Device
'SIP/136' changed to state '0'<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'>2004-12-08 15:21:50 DEBUG[4650]: Hangup:
channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1<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'>2004-12-08 15:21:50 DEBUG[4650]: disabled
echo cancellation on channel 1<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'>2004-12-08 15:21:50 DEBUG[4650]: Set
option TDD MODE, value: OFF(0) on Zap/1-1<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>&nbsp;</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'>Thanks,<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>&nbsp;</o:p></span></font></p>

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<p class=MsoNormal><st1:PersonName w:st="on"><font size=2 color=navy
 face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Jared
 Armstrong</span></font></st1:PersonName><font color=navy><span
style='color:navy'><o:p></o:p></span></font></p>

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face="Times New Roman"><span style='font-size:12.0pt'>

<hr size=2 width="100%" align=center tabindex=-1>

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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'> <st1:PersonName
w:st="on">Jared Armstrong</st1:PersonName> <br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, December 08, 2004
1:12 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">'asterisk-users@lists.digium.com'</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: Dropping Calls,
irregular interval no logs</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>&nbsp;</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'>Ok, So I removed the Qualify=2000 from my
sip.conf file and set NAT=no and I changed the connection negotiation in my
Polycom config to 3600 (default) and now I haven&#8217;t had a dropped call
yet. Does anyone know which of these if any might be the cause or if there is
something else I am still overlooking?<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>&nbsp;</o:p></span></font></p>

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<p class=MsoNormal><st1:PersonName w:st="on"><font size=2 color=navy
 face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Jared
 Armstrong</span></font></st1:PersonName><font color=navy><span
style='color:navy'><o:p></o:p></span></font></p>

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