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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'>Just a &#8220;shot in the dark&#8221; but
could MOH be choking on the &#8220;long file names&#8221;?  (does it work on
fred_chopin_pol_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>

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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>Dan Saul<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, September 16,
2009 4:18 PM<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> [asterisk-users] Music on
Hold</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 style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Hi,<br>
<br>
I have trouble getting MOH to work after an upgrade from asterisk 1.4 to
1.6.1.4. The call goes on hold, MOH is started, and then stops right away.<br>
<br>
Here are the files both of type .raw:<br>
<br>
Tsunami*CLI&gt; moh show files<br>
Class: default<br>
&nbsp;&nbsp;&nbsp; File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises
Op. 40-2<br>
&nbsp;&nbsp;&nbsp; File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises
Op. 40-1<br>
<br>
These files were generated by SoX:<br>
Channels&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 1<br>
Sample Rate&nbsp;&nbsp;&nbsp; : 8000<br>
Precision&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 16-bit<br>
Sample Encoding: 16-bit Signed Integer PCM<br>
Endian Type&nbsp;&nbsp;&nbsp; : little<br>
Reverse Nibbles: no<br>
Reverse Bits&nbsp;&nbsp; : no<br>
Comment&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 'Processed by SoX'<br>
<br>
This prints in the asterisk console when you attempt to put someone in hold:<br>
<br>
&nbsp;&nbsp;&nbsp; -- Started music on hold, class 'default', on
SIP/link2voip-sw1-02477668<br>
&nbsp;&nbsp;&nbsp; -- Stopped music on hold on SIP/link2voip-sw1-02477668<br>
<br>
No errors are printed, however the other side just hears silence.<br>
<br>
Here is the full debug output (asterisk -rvvvvv):<br>
<br>
&nbsp;== Using SIP RTP CoS mark 5<br>
&nbsp;&nbsp;&nbsp; -- Executing [xxxxxxx@phones:1]
Goto(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;, &quot;1xxxxxxxxxx,1&quot;) in
new stack<br>
&nbsp;&nbsp;&nbsp; -- Goto (phones,1xxxxxxxxxx,1)<br>
&nbsp;&nbsp;&nbsp; -- Executing [1xxxxxxxxxx@phones:1]
MSet(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;, &quot;oldcidnum=0&quot;) in
new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [1xxxxxxxxxx@phones:2]
MSet(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;,
&quot;CALLERID(name)=&quot;&quot;&quot;) in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [1xxxxxxxxxx@phones:3]
MSet(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;,
&quot;CALLERID(num)=xxxxxxxxxx&quot;) in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [1xxxxxxxxxx@phones:4]
Monitor(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;, &quot;wav,/tmp/out 0
2009-09-17 03h 04m 51s CST xxxxxxxxxx,m&quot;) in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [1xxxxxxxxxx@phones:5]
Gosub(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;, &quot;ExternalDial,s,1(1xxxxxxxxxx)&quot;)
in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [s@ExternalDial:1]
MSet(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;,
&quot;LOCAL(num)=1xxxxxxxxxx&quot;) in new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [s@ExternalDial:2]
MSet(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;, &quot;~~EXTEN~~=s&quot;) in
new stack<br>
&nbsp;&nbsp;&nbsp; -- Executing [s@ExternalDial:3]
Dial(&quot;SIP/ATA-xxxxxxxxxx-L1-024b6d88&quot;,
&quot;SIP/1xxxxxxxxxx@link2voip-sw1,120&quot;) in new stack<br>
&nbsp; == Using SIP RTP CoS mark 5<br>
&nbsp;&nbsp;&nbsp; -- Called 1xxxxxxxxxx@link2voip-sw1<br>
&nbsp;&nbsp;&nbsp; -- SIP/link2voip-sw1-02477668 is making progress passing it
to SIP/ATA-xxxxxxxxxx-L1-024b6d88<br>
&nbsp;&nbsp;&nbsp; -- SIP/link2voip-sw1-02477668 answered
SIP/ATA-xxxxxxxxxx-L1-024b6d88<br>
&nbsp;&nbsp;&nbsp; -- Started music on hold, class 'default', on
SIP/link2voip-sw1-02477668<br>
&nbsp;&nbsp;&nbsp; -- Stopped music on hold on SIP/link2voip-sw1-02477668<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; doing dnsmgr_lookup for '<a
href="http://sip.ca2.link2voip.com">sip.ca2.link2voip.com</a>'<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; doing dnsmgr_lookup for '<a
href="http://sip.ca1.link2voip.com">sip.ca1.link2voip.com</a>'<br>
&nbsp; == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xxxxxxxxxx-L1-024b6d88'<br>
<br>
Any thoughts or ideas? If there were an error I could work on solving that, but
there is none... Thanks.<o:p></o:p></span></font></p>

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