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

[Tzafrir Cohen]

Hi,

Is the issue still reproduced? If so:

On Wed, Mar 31, 2010 at 11:22:39AM -0400, Alex . wrote:
> 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...

Can you check in the logs (/var/log/asterisk/messages) if the call is
hung up when you stop hearing the audio? Or later?

If no extra useful messages in the log, please enable the 'full' log in
/etc/asterisk/logger.conf (apply with 'logger reload'), set debug level
to 2, and try calling in. You can skip the SIP debug at this stage,
though - it mostly adds noise right now.

Can you provide the relevant lines from /var/log/asterisk/full ?

> 
> cheers!
> 
> 
> 
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> Pkg-voip-maintainers at lists.alioth.debian.org
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-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir