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