[asterisk-users] getting real sip status after dial

[Eric Wieling]

My actual dialplan hangupcause handling is in AGI and AEL, so the 
dialplan below has not been tested.

Don't use HANGUPCAUSE_KEYS, the order of the channels varies.  Save the 
channel name while the dialplan is in the predial handler like below. 
MASTER_CHANNEL is used to avoid silly issues with the scope of the 
out_chan variable.

[test]
exten => _X.,1,Set(CHANNEL(hangup_handler_push)=caller_hangup,s,1)
  same => n,Dial(SIP/my-peer/12125551212,30,b(test_pre_dial^s^1))

[test_pre_dial]
; while in the handler save the channel
exten => _X.,1,Set(MASTER_CHANNEL(out_chan)=${CHANNEL{name})
  same => n,Return

You should be able to use the caller hangup handler to get tech 
hangupcauses like "SIP 480 Temporarily Unavailable".

[caller_hangup]
exten => 
s,1,Noop(HANGUPCAUSE(${MASTER_CHANNEL(out_chan)},tech)='${HANGUPCAUSE(${MASTER_CHANNEL(out_chan)},tech)}')
exten => n,Return


On 06/09/2018 03:10 PM, Khalil Khamlichi wrote:
> Thanks for your response Eric,
> 
> Here is some testing code, as you can see ${HANGUPCAUSE(${ARG1},tech)}
> is empty if number is not found (HANGUPCAUSE=1) or if sip request
> times-out (HANGUPCAUSE=0) (a dead far end for example) so I had to
> check value HANGUPCAUSE and filter out zero and one before moving on.
> where all I need is simply the sip code returned from the server that
> even softphones are able to give me.
> 
> 
> [autodial_out]
> exten => _X.,1,NoOP( testing manager dial out )
>   same => n,Dial(PJSIP/${EXTEN}@${TRUNK},25,b(autodial_out^setup_hup_handler^1)g)
>   same => n,Hangup()
> 
> 
> exten => setup_hup_handler,1,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1(${CHANNEL}))
>   same => n,Return()
> 
> 
> [hdlr1]
> exten => s,1,NoOp(START==============================================================================)
>   same => n,Set(HANGUPCAUSE_STRING=${HANGUPCAUSE_KEYS()})
>   same => n,Set(ASTcause=${HANGUPCAUSE})
>   same => n,Set(GO=Found)
>   same => n,GotoIf($[${ASTcause}=0]?NotFound0:Check2)
>   same => n(Check2),GotoIf($[${ASTcause}=1]?NotFound1:Found)
>   same => n(Found),Set(SIPcause=${HANGUPCAUSE(${ARG1},tech)})
>   same => n,Goto(End)
>   same => n(NotFound0),Set(SIPcause=SIP 500 Server Internal Error)
>   same => n,Goto(End)
>   same => n(NotFound1),Set(SIPcause=SIP 404 Not Found)
>   same => n,Goto(End)
>   same => n(End),Set(============================ ${SIPcause} /
> ${ASTcause} =======================END)
>   same => n,Return()
> 
> 
> On Sat, Jun 9, 2018 at 7:02 PM Eric Wieling <ewieling at nyigc.com> wrote:
>>
>> I think HANGUPCAUSE is channel agnostic.
>>
>> See: core show function HANGUPCAUSE
>>
>> Some thing like this IIRC:
>>       Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)})
>>
>> Remember the incoming leg of the call and the outgoing leg of the call
>> are different channels.  Make sure you are giving HANGUPCAUSE the
>> correct channel.
>>
>> On 06/09/2018 02:01 PM, Khalil Khamlichi wrote:
>>> It seems very weird to me that we cannot access sip code of a call
>>> from pjsip which information is actually returned from the provider,
>>> so it is available to asterisk, why does asterisk hide it ?
>>> On Sat, Jun 9, 2018 at 5:08 PM Khalil Khamlichi
>>> <khamlichi.khalil at gmail.com> wrote:
>>>>
>>>> Hi,
>>>>
>>>> Is there any way I can get exact sip status from pjsip after a dial ?
>>>> or all we can
>>>> get is asterisk hangup causes ?
>>>>
>>>> Thanks in advance.
>>>>
>>>> KKh
>>>
>>
>> --
>> http://help.nyigc.net/

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