Administrator TOOTAI
2015-Mar-23 15:45 UTC
[asterisk-users] Question about hangup - Asterisk v11.15.0
Hello,
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
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
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 =
n,Set(__phonesToRing=SIP/118&SIP/119&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/130&SIP/132)
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
SIP/bero_trunk-000043d2
[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
stack
Thanks for any hint
--
Daniel