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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Thanks Danny. I tried that but all that did is make it so when I call the site, I get hold music instead of ringing. Still has no affect on the call transfer MOH. :/<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Interestingly, the music is playing for about 3-5 seconds before stopping during the transfer.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">I’ve built all of my phone servers the same at my sites and I’m still sorta green on some of this stuff. When doing a call transfer, is it using the macro-stdexten or does
it go to the IVR dial plan? Because the entry you noted is in the IVR dialplan, not the macro-stdexten.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Here’s my macro-stdexten:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">[macro-stdexten]
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,1,wait(1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,2,Dial(${ARG2},20)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,3,Goto(s-${DIALSTATUS},1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,4,Dial(${ARG2},15) ; Ring phone for 15 seconds<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to voicemail w/ busy announce
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s-BUSY,2,Hangup<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable, send to voicemail<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s-NOANSWER,2,Hangup<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => a,1,VoicemailMain(${CALLERIDNUM})<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => a,2,Hangup<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Here’s my IVR Dialplan:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,1,Set(TIMEOUT(digit)=4)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Wait(1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Answer<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Dial(SIP/1021,20)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Wait(1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Dial(SIP/6909,15)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Dial(SIP/6904,15)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Voicemail(1021,u)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">exten => s,n,Hangup<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Note: I removed the “,m” because it was only affecting the new incoming calls.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif"">Here’s my CLI output:<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [1021@from-PRI:1] Goto("SIP/gw1-00000066", "ivr-boi-ntc,s,1") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Goto (ivr-boi-ntc,s,1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@ivr-boi-ntc:1] GotoIfTime("SIP/gw1-00000066", "07:00-17:00,mon-fri,*,*?ivr-boi-ntc-day,s,1") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Goto (ivr-boi-ntc-day,s,1)<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@ivr-boi-ntc-day:1] Set("SIP/gw1-00000066", "TIMEOUT(digit)=4") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Digit timeout set to 4.000<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@ivr-boi-ntc-day:2] Wait("SIP/gw1-00000066", "1") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-00000066", "") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@ivr-boi-ntc-day:4] Dial("SIP/gw1-00000066", "SIP/1021,20,m") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Called 1021<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Started music on hold, class 'default', on SIP/gw1-00000066<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/1021-00000067 is ringing<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/1021-00000067 answered SIP/gw1-00000066<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Stopped music on hold on SIP/gw1-00000066<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Packet2Packet bridging SIP/gw1-00000066 and SIP/1021-00000067<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [12086597642@from-sip:1] Set("SIP/6908-00000068", "CALLERID(num)=2084331021") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [12086597642@from-sip:2] Dial("SIP/6908-00000068", "SIP/gw1/12086597642,60") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Called gw1/12086597642<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/gw1-00000069 is ringing<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Started music on hold, class 'default', on SIP/gw1-00000066<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/gw1-00000069 is making progress passing it to SIP/6908-00000068<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [6911@from-sip:1] Macro("SIP/1021-0000006a", "stdexten,6911,sip/6911") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@macro-stdexten:1] Wait("SIP/1021-0000006a", "1") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Executing [s@macro-stdexten:2] Dial("SIP/1021-0000006a", "sip/6911,20") in new stack<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Using SIP RTP CoS mark 5<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Called 6911<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/6911-0000006b is ringing<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Stopped music on hold on SIP/gw1-00000066<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Spawn extension (ivr-boi-ntc-day, s, 4) exited non-zero on 'SIP/1021-0000006a<ZOMBIE>'<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/gw1-00000066' in macro 'stdexten'<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> == Spawn extension (from-sip, 6911, 1) exited non-zero on 'SIP/gw1-00000066'<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- SIP/gw1-00000069 answered SIP/6908-00000068<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif""> -- Packet2Packet bridging SIP/6908-00000068 and SIP/gw1-00000069<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D">Thanks,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<div>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";
color:#333333">Kevin Oravits
</span></b><span style="font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D"><o:p></o:p></span></p>
</div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D"><o:p> </o:p></span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Danny Nicholas [mailto:danny@debsinc.com]
<br>
<b>Sent:</b> Tuesday, August 30, 2011 12:40 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] MOH making calls appear hung up<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Your Dial command stops the MOH – if the command were Dial(SIP/1021,20,m) the music would continue until connected or timed-out.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>From:</b> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b>On Behalf Of </b>Kevin Oravits<br>
<b>Sent:</b> Tuesday, August 30, 2011 2:37 PM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] MOH making calls appear hung up<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I noticed the CLI shows that the music on hold actually stops for some reason?<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Here’s the output of my CLI:<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Verbosity is at least 28<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-000005d6", "") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-000005d6", "1") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-000005d6", "SIP/1021,20") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Using SIP RTP CoS mark 5<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Called 1021<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- SIP/1021-000005d7 is ringing<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- SIP/1021-000005d7 answered SIP/gw1-000005d6<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Packet2Packet bridging SIP/gw1-000005d6 and SIP/1021-000005d7<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Started music on hold, class 'default', on SIP/gw1-000005d6<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Using SIP RTP CoS mark 5<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [6937@from-sip:1] Macro("SIP/1021-000005d8", "stdexten,6937,sip/6937") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@macro-stdexten:1] Wait("SIP/1021-000005d8", "1") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@macro-stdexten:2] Dial("SIP/1021-000005d8", "sip/6937,20") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Using SIP RTP CoS mark 5<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Called 6937<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- SIP/6937-000005d9 is ringing<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Stopped music on hold on SIP/gw1-000005d6<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on 'SIP/1021-000005d8<ZOMBIE>'<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Nobody picked up in 20000 ms<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s@macro-stdexten:3] Goto("SIP/gw1-000005d6", "s-NOANSWER,1") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Goto (macro-stdexten,s-NOANSWER,1)<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-000005d6", "6937,u") in new stack<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> -- <SIP/gw1-000005d6> Playing '/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/gw1-000005d6' in macro 'stdexten'<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> == Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-000005d6'<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thanks!<o:p></o:p></p>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Kevin Oravits
</b><o:p></o:p></p>
</div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>From:</b> Danny Nicholas [mailto:danny@debsinc.com]
<br>
<b>Sent:</b> Tuesday, August 30, 2011 11:56 AM<br>
<b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br>
<b>Subject:</b> Re: [asterisk-users] MOH making calls appear hung up<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">It seems a reasonable likelihood that your moh at the offending site does not match the codec of the call (IE your MOH is wav and your call codec is SLIN). Set your verbosity and
debug up to 15 and try a call to verify this.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>From:</b> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b>On Behalf Of </b>Kevin Oravits<br>
<b>Sent:</b> Tuesday, August 30, 2011 1:53 PM<br>
<b>To:</b> 'asterisk-users@lists.digium.com'<br>
<b>Subject:</b> [asterisk-users] MOH making calls appear hung up<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Greetings,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I’m have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we’re having an issue where when a call is transferred, the MOH is not playing
and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I have researched online and even consulted our 3<sup>rd</sup> party vendor but no one seems to know how to fix it.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Anyone have any advice? Any help would be appreciated.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thanks,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><o:p> </o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>Stivaro</b><o:p></o:p></p>
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