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<p class=MsoNormal>Hello All,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I’m having an issue where asterisk doesn’t hear
any audio after transferring to voicemail. Here is the dial plan and console
output. <o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>DIAL PLAN<o:p></o:p></p>
<p class=MsoNormal>[voicepulse-in]<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,1,NoOp(Incoming call from
VoicePulse)<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,2,Ringing<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,3,Wait(1)<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,4,Dial(SIP/1337,20)<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,5,VoiceMail(1337)<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,6,Wait(1)<o:p></o:p></p>
<p class=MsoNormal>exten => _14259491337,7,HangUp<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>CONSOLE OUTPUT<o:p></o:p></p>
<p class=MsoNormal>-- Executing [14259491337@voicepulse-in:4]
Dial("SIP/<vpuser>-081d2800", "SIP/1337|20") in new
stack<o:p></o:p></p>
<p class=MsoNormal> -- Called 1337<o:p></o:p></p>
<p class=MsoNormal> -- SIP/1337-081cfce0 is ringing<o:p></o:p></p>
<p class=MsoNormal> -- Nobody picked up in 20000 ms<o:p></o:p></p>
<p class=MsoNormal> -- Executing
[14259491337@voicepulse-in:5] VoiceMail("SIP/<vpuser>-081d2800",
"1337") in new stack<o:p></o:p></p>
<p class=MsoNormal> -- <SIP/<vpuser>-081d2800>
Playing 'vm-intro' (language 'en')<o:p></o:p></p>
<p class=MsoNormal> -- <SIP/<vpuser>-081d2800>
Playing 'beep' (language 'en')<o:p></o:p></p>
<p class=MsoNormal> -- Recording the message<o:p></o:p></p>
<p class=MsoNormal> -- x=0, open writing:
/var/spool/asterisk/voicemail/default/1337/tmp/KzD4A1 format: wav, 0x8184358<o:p></o:p></p>
<p class=MsoNormal>[Oct 31 08:21:02] WARNING[22354]: app.c:602
__ast_play_and_record: No audio available on SIP/<vpuser>-081d2800??<o:p></o:p></p>
<p class=MsoNormal> -- User hung up<o:p></o:p></p>
<p class=MsoNormal>[Oct 31 08:21:02] NOTICE[22354]: pbx.c:1631
pbx_substitute_variables_helper_full: Error in extension logic (missing '}')<o:p></o:p></p>
<p class=MsoNormal> == Spawn extension (voicepulse-in, 14259491337, 5)
exited non-zero on 'SIP/<vpuser>-081d2800'<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Any help would be greatly appreciated!<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Thanks,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Jeremy Phillips<br>
<span style='font-size:9.0pt;color:gray'>M: 540.322.7980 | T: 425.949.1337 | B:
http://jeremyphillips.org</span><span style='font-size:9.0pt'><o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
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