Bug#575465: asterisk: voicemail hangs in vm-intro before the beep unless "s" flag is used

[Alex .]

Thanks for the tip, i tried to fix the Voicemail position but realised
it was actually going into NOANSWER.
So i did a new test that looks something like this:

I still get the same result though, first call i get farther into the
message than following ones.
(that's after restarting asterisk completely from the init script)

[inbound]
exten => 5145551212,1,Dial(SIP/spaa&SIP/spab&SIP/spac&SIP/spad&SIP/n9,20,wt)
exten => 5145551212,n,Goto(in5145551212-${DIALSTATUS},1)
exten => 5145551212,n,Hangup

exten => in5145551212-BUSY,1,Voicemail(2102,b)
exten => in5145551212-BUSY,n,Hangup(17)

exten => in5145551212-CONGESTION,1,Voicemail(2102)
exten => in5145551212-CONGESTION,n,Hangup(3)

exten => in5145551212-CHANUNAVAIL,1,Voicemail(2102,u)
exten => in5145551212-CHANUNAVAIL,n,Hangup()

exten => in5145551212-NOANSWER,1,Voicemail(2102,u)
exten => in5145551212-NOANSWER,n,Hangup(16)

exten => _in5145551212-.,1,Hangup(16)


(with level 10 verbosity)
-------------------- START FIRST TRY / CALL, AFTER RESTART OF ASTERISK
-----------------------

       > Saved useragent "Telepathy-SofiaSIP/0.5.18.1
sofia-sip/1.12.10devel" for peer n9
    -- Accepting UNAUTHENTICATED call from 206.191.37.138:
       > requested format = ulaw,
       > requested prefs = (ulaw|gsm),
       > actual format = gsm,
       > host prefs = (),
       > priority = caller
    -- Executing [5145551212 at unlimitel-inbound:1]
Dial("IAX2/206.191.37.138:4569-13",
"SIP/spaa&SIP/spab&SIP/spac&SIP/spad&SIP/n9,20,wt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
    -- Called spaa
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
    -- Called spab
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
    -- Called spac
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
    -- Called spad
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
[Mar 31 10:58:36] WARNING[6383]: app_dial.c:1745 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
    -- Called n9
    -- SIP/spaa-00000000 is ringing
    -- SIP/spac-00000002 is ringing
    -- SIP/spab-00000001 is ringing
    -- SIP/spad-00000003 is ringing
    -- SIP/n9-00000004 is ringing
    -- Got SIP response 480 "Terminated" back from 192.168.1.129
    -- SIP/n9-00000004 is circuit-busy
    -- Nobody picked up in 2000 ms
    -- Executing [5145551212 at unlimitel-inbound:2]
Goto("IAX2/206.191.37.138:4569-13", "in5145551212-NOANSWER,1") in new
stack
    -- Goto (unlimitel-inbound,in5145551212-NOANSWER,1)
    -- Executing [in5145551212-NOANSWER at unlimitel-inbound:1]
VoiceMail("IAX2/206.191.37.138:4569-13", "2102,u") in new stack
    -- <IAX2/206.191.37.138:4569-13> Playing
'/var/spool/asterisk/voicemail/default/2102/unavail.gsm' (language
'fr')
    -- <IAX2/206.191.37.138:4569-13> Playing 'vm-intro.gsm' (language 'fr')
  == Spawn extension (unlimitel-inbound, in5145551212-NOANSWER, 1)
exited non-zero on 'IAX2/206.191.37.138:4569-13'
    -- Hungup 'IAX2/206.191.37.138:4569-13'

-------------------- END FIRST TRY / CALL, AFTER RESTART OF ASTERISK
-----------------------

The second call looks the same but cuts after "Bonjour" (welcome)
instead of farther into the voicemail message.
In both case I didn't get any beep...

cheers!