Cannot leave voicemail for any extension. *RESOLVED*


#1

I’m sorry to say that I have beat my head against this and have googled, looked in the forums, and read the documentation. I am new to Asterisk however, I have everything working with the exception of this little problem. Some direction on this would be welcomed.

Asterisk Version: 1.8.5.0 (source built)
Asterisk GUI Version: 2.0 (source built)
Linux Distribution: Ubuntu 10.04 LTS (64-bit)
Kernel Version: 2.6.32-28 x86_64

A call coming in from another extension or a trunk when triggered to go to voicemail - namely due to NO ANSWER will play the user’s voicemail message and then hang up the call. The error occurs after the Goto (default,s,1) is called where Asterisk reports that ‘s’ is not a valid extension in the context of ‘default’. I cannot figure out in the GUI how to look at where the default is configured (if at all), and if so, what I need to put in its place.

I have configured this system nearly entirely by use of the Asterisk-GUI v2.0

The CLI output is as follows:

    -- Executing [6001@DLPN_Users:1] Macro("SIP/6000-00000014", "stdexten,6001,SIP/6001") in new stack
    -- Executing [s@macro-stdexten:1] Set("SIP/6000-00000014", "__DYNAMIC_FEATURES=") in new stack
    -- Executing [s@macro-stdexten:2] Set("SIP/6000-00000014", "ORIG_ARG1=6001") in new stack
    -- Executing [s@macro-stdexten:3] GotoIf("SIP/6000-00000014", "0?6:4") in new stack
    -- Goto (macro-stdexten,s,4)
    -- Executing [s@macro-stdexten:4] Dial("SIP/6000-00000014", "SIP/6001,20,") in new stack
[Aug 14 17:46:58] WARNING[9262]: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-stdexten:5] Goto("SIP/6000-00000014", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-stdexten:1] Goto("SIP/6000-00000014", "s-NOANSWER,1") in new stack
    -- Goto (macro-stdexten,s-NOANSWER,1)
    -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/6000-00000014", "6001,u") in new stack
[Aug 14 17:46:58] WARNING[9262]: res_rtp_asterisk.c:2044 ast_rtp_read: RTP Read too short
    -- <SIP/6000-00000014> Playing 'vm-theperson.gsm' (language 'en')
    -- <SIP/6000-00000014> Playing 'digits/6.gsm' (language 'en')
    -- <SIP/6000-00000014> Playing 'digits/0.gsm' (language 'en')
    -- <SIP/6000-00000014> Playing 'digits/0.gsm' (language 'en')
    -- <SIP/6000-00000014> Playing 'digits/1.gsm' (language 'en')
    -- <SIP/6000-00000014> Playing 'vm-isunavail.gsm' (language 'en')
    -- Executing [s-NOANSWER@macro-stdexten:2] Goto("SIP/6000-00000014", "default,s,1") in new stack
    -- Goto (default,s,1)
  == Channel 'SIP/6000-00000014' jumping out of macro 'stdexten'
[Aug 14 17:47:04] WARNING[9262]: pbx.c:5139 __ast_pbx_run: Channel 'SIP/6000-00000014' sent into invalid extension 's' in context 'default', but no invalid handler

Is the context ‘default’ spoken of here something misconfigured in voicemail.conf, extensions.conf, or elsewhere? I can include additional configuration files, but wasn’t sure which would be appropriate at this point.


#2

default is the last resort context. However, this is an explicit jump to default, not a fallback because the extension wasn’t found. s is also the default extension, used as a last resort in some cases.

Voicemail has already gone wrong before the Goto, and it looks like the Goto is an attempt to basically put the call back into a dial tone like state, so that the caller can try another number.

However, as this is a GUI, you really need to ask on forum for that GUI.


#3

In my particular installation I have found that off of the original install if I go into the Asterisk-GUI and set a voicemail extension for checking messages the system screws up the voicemail system. I have not taken the time to go through the actual configuration files to see what it does, but I’ll close this thread as resolved and move the bug over to the Asterisk-GUI forum. Thanks for your reply.