[asterisk-users] Question about hangup - Asterisk v11.15.0

Administrator TOOTAI admin at tootai.net
Mon Mar 23 10:45:31 CDT 2015


on previous versions of asterisk, extension h and H make us know who 
ended a call (caller or callee). In the last * versions, seems that only 
h extension is used, as stated here 

In the last versions, how do we know which end terminate a call (SIP, 
ISDN, Analog, ...) in h extension ? Will the 
${HASH(SIP_CAUSE,${CDR(dstchannel)})} give the information ?

We also face a strange behavior: we are ringing few phones (~10) and 
sometimes, once the call get answered, we see that 2~3 seconds after 
this, music on hold is started on the channel! And 20 seconds after, the 
call is terminated without that any party hanged up :-(

It's a Elastix 2.5 installation, we thought that problem could came from 
Elastix so we set our own dialplan for incoming calls:

  same = 
  same = n(startRing),Answer()
  same = n,Dial(${phonesToRing},,it)                     ;no voicemail 
or forward => ring indefenitely
  same = n,Hangup

Incoming call give for instance in logs:

[2015-03-23 11:07:20] VERBOSE[1342][C-00000e85] app_dial.c:     -- 
SIP/126-000043d8 is ringing
[2015-03-23 11:07:21] VERBOSE[1342][C-00000e85] app_dial.c:     -- 
SIP/118-000043d3 connected line has changed. Saving it until answer for 
[2015-03-23 11:07:21] VERBOSE[1342][C-00000e85] app_dial.c:     -- 
SIP/118-000043d3 answered SIP/bero_trunk-000043d2
[2015-03-23 11:07:25] VERBOSE[1342][C-00000e85] res_musiconhold.c:     
-- Started music on hold, class 'default', on SIP/bero_trunk-000043d2
[2015-03-23 11:07:27] VERBOSE[1342][C-00000e85] res_musiconhold.c:     
-- Stopped music on hold on SIP/bero_trunk-000043d2
[2015-03-23 11:07:41] VERBOSE[1342][C-00000e85] pbx.c:     -- Executing 
[h at from-trunk:1] Macro("SIP/bero_trunk-000043d2", "hangupcall,") in new 

Thanks for any hint


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