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.