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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>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></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><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"'> 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></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>I noticed the CLI shows that the music on hold actually stops for some reason?<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"'>Here’s the output of my CLI:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Verbosity is at least 28<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/gw1-000005d6", "") 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] Wait("SIP/gw1-000005d6", "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:5] Dial("SIP/gw1-000005d6", "SIP/1021,20") 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 1021<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- SIP/1021-000005d7 is ringing<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- SIP/1021-000005d7 answered SIP/gw1-000005d6<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- Packet2Packet bridging SIP/gw1-000005d6 and SIP/1021-000005d7<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/gw1-000005d6<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 [6937@from-sip:1] Macro("SIP/1021-000005d8", "stdexten,6937,sip/6937") 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/1021-000005d8", "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/1021-000005d8", "sip/6937,20") 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 6937<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- SIP/6937-000005d9 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/gw1-000005d6<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, 5) exited non-zero on 'SIP/1021-000005d8<ZOMBIE>'<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- Nobody picked up in 20000 ms<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- Executing [s@macro-stdexten:3] Goto("SIP/gw1-000005d6", "s-NOANSWER,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 (macro-stdexten,s-NOANSWER,1)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-000005d6", "6937,u") in new stack<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> -- <SIP/gw1-000005d6> Playing '/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en')<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-NOANSWER, 1) exited non-zero on 'SIP/gw1-000005d6' 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, 6937, 1) exited non-zero on 'SIP/gw1-000005d6'<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"'>Thanks!<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 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></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'>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></div></div></body></html>