<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">Hi All,</span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; "><br></span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">-My asterisk will not save voicemail greetings when you call in and record them. </span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">-It also will not save voicemail messages after emailing them,even though delete=no. </span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">-Folder permissions are fine, no errors in asterisk cli. </span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">-If i go into /var/spool/asterisk/voicemail/default/200 and touch unavail.wav, and then call in and record new unavail message, unavail.wav disappears?</span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; "><br></span></font></p><p style="margin-top: 0px; margin-right: 0px; margin-bottom: 2px; margin-left: 10px; text-indent: -10px; font: normal normal normal 12px/normal Helvetica; "><font class="Apple-style-span" face="Verdana" size="4"><span class="Apple-style-span" style="font-size: 14px; ">Can anyone help point me towards any possible info to fix this, i'm stumped and losing hair!</span></font></p><div><br></div><div>I am running</div><div><br></div><div>FreePBX 2.5.1</div><div>Asterisk 1.4.23.1</div><div>CentOS 5.3</div><div><br></div><div>CLI Output during my attempt to call in and record a greeting:</div><div><br></div><div><div> -- Executing [*97@from-internal:4] Macro("SIP/200-00fd3150", "get-vmcontext|200") in new stack</div><div> -- Executing [s@macro-get-vmcontext:1] Set40m("SIP/200-00fd3150", "VMCONTEXT=default") in new stack</div><div> -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/200-00fd3150", "0?200:300") in new stack</div><div> -- Goto (macro-get-vmcontext,s,300)</div><div> -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/200-00fd3150", "") in new stack</div><div> -- Executing [*97@from-internal:5] MailboxExists37;40m("SIP/200-00fd3150", "200@default") in new stack</div><div> -- Executing [*97@from-internal:6] GotoIf("SIP/200-00fd3150", "1?mbexist") in new stack</div><div> -- Goto (from-internal,*97,106)</div><div> -- Executing [*97@from-internal:106] VoiceMailMain("SIP/200-00fd3150", "200@default") in new stack</div><div> -- <SIP/200-00fd3150> Playing 'vm-password' (language 'en')</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> -- <SIP/200-00fd3150> Playing 'vm-youhave' (language 'en')</div><div> -- <SIP/200-00fd3150> Playing 'vm-no' (language 'en')</div><div> -- <SIP/200-00fd3150> Playing 'vm-messages' (language 'en')</div><div> -- <SIP/200-00fd3150> Playing 'vm-opts' (language 'en')</div><div> -- <SIP/200-00fd3150> Playing 'vm-options' (language 'en')</div><div> -- Recording the message</div><div> -- <SIP/200-00fd3150> Playing 'vm-rec-unv' (language 'en')</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> -- <SIP/200-00fd3150> Playing 'beep' (language 'en')</div><div> -- x=0, open writing: /var/spool/asterisk/voicemail/default/200/unavail.tmp format: wav49, 0x97d788</div><div> -- x=1, open writing: /var/spool/asterisk/voicemail/default/200/unavail.tmp format: gsm, 0x9eecd8</div><div> -- x=2, open writing: /var/spool/asterisk/voicemail/default/200/unavail.tmp format: wav, 0xa4f778</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> -- User ended message by pressing #</div><div> -- <SIP/200-00fd3150> Playing 'auth-thankyou' (language 'en')</div><div> -- <SIP/200-00fd3150> Playing 'vm-review' (language 'en')</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> -- Saving message as is</div><div> -- <SIP/200-00fd3150> Playing 'vm-msgsaved' (language 'en')</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> -- <SIP/200-00fd3150> Playing 'vm-options' (language 'en')</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> == Parsing '/etc/asterisk/manager.conf': Found</div><div> == Parsing '/etc/asterisk/manager_additional.conf': Found</div><div> == Parsing '/etc/asterisk/manager_custom.conf': Found</div><div> == Manager 'admin' logged on from 127.0.0.1</div><div> == Manager 'admin' logged off from 127.0.0.1</div><div> == Spawn extension (from-internal, *97, 106) exited non-zero on 'SIP/200-00fd3150'</div><div> -- Executing [h@from-internal:1] Macro("SIP/200-00fd3150", "hangupcall") in new stack</div><div> -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/200-00fd3150", "w") in new stack</div><div> -- Executing [s@macro-hangupcall:2] NoCDR("SIP/200-00fd3150", "") in new stack</div><div> -- Executing [s@macro-hangupcall:3] GotoIf("SIP/200-00fd3150", "1?skiprg") in new stack</div><div> -- Goto (macro-hangupcall,s,6)</div><div> -- Executing [s@macro-hangupcall:6] GotoIf("SIP/200-00fd3150", "1?skipblkvm") in new stack</div><div> -- Goto (macro-hangupcall,s,9)</div><div> -- Executing [s@macro-hangupcall:9] GotoIf("SIP/200-00fd3150", "1?theend") in new stack</div><div> -- Goto (macro-hangupcall,s,11)</div><div> -- Executing [s@macro-hangupcall:11] Hangup("SIP/200-00fd3150", "") in new stack</div><div> == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-00fd3150' in macro 'hangupcall'</div><div> == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-00fd3150'</div><div><br></div><div><br></div><div>Here is voicemail.conf:</div><div><br></div><div><div>[general]</div><div><br></div><div>format = wav49|gsm|wav</div><div><br></div><div>serveremail = <a href="mailto:voicemail@edited.com">voicemail@edited.com</a></div><div>fromstring = Voicemail System</div><div><br></div><div>attach = yes</div><div><br></div><div>skipms = 3000</div><div><br></div><div>maxsilence = 10</div><div><br></div><div>silencethreshold = 128</div><div><br></div><div>maxlogins = 3</div><div><br></div><div>emailsubject = Message from ${VM_CALLERID} in mailbox ${VM_MAILBOX}</div><div><br></div><div>emailbody = Dear ${VM_NAME}:\n\nThis is a message to inform you of a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance.\n\n</div><div><br></div><div>emaildateformat = %A, %B %d, %Y at %r</div><div><br></div><div>[zonemessages]</div><div>eastern = America/New_York|'vm-received' Q 'digits/at' IMp</div><div>central = America/Chicago|'vm-received' Q 'digits/at' IMp</div><div>central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'</div><div>military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'</div><div>european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM</div><div><br></div><div>[default]</div><div>202 => 1234,Chad Edited,<a href="mailto:chad@edited.com">chad@edited.com</a>,,attach=yes|saycid=no|envelope=yes|delete=no</div><div>300 => 1234,Don Edited,<a href="mailto:donextreme@edited.com">donextreme@edited.com</a>,,attach=yes|saycid=no|envelope=yes|delete=no</div><div>201 => 1234,Danica Edited,<a href="mailto:danica@edited.com">danica@edited.com</a>,,attach=yes|saycid=no|envelope=yes|delete=no</div><div>200 => 3816,Camron Edited,<a href="mailto:cam@edited.com">cam@edited.com</a>,,attach=yes|saycid=no|envelope=yes|delete=no</div></div><div><br></div><div>Let me know if any additional information is needed.</div></div><div><br></div><div><div> <span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Verdana; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0; "><div style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>Respectfully,</div><div><br></div><div>Dr. Kenneth Noisewater, Phd</div><div><br></div><div><br></div></div></span><br class="Apple-interchange-newline"> </div><br><div><div>On Apr 11, 2009, at 5:50 AM, Doug Lytle wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>Dr. Kenneth Noisewater wrote:<br><blockquote type="cite"><br></blockquote><blockquote type="cite">-Asterisk and FreePBX source installs on CentOS 5.4<br></blockquote><blockquote type="cite"><br></blockquote><br>Without version numbers and console output and samples of your dialplan, <br>it'g going to make it very difficult to help.<br><br>Doug<br><br><br>-- <br>Ben Franklin quote:<br><br>"Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."<br><br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></div></blockquote></div><br></div></body></html>