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/
Khalil Khamlichi
2018-Jun-09 19:10 UTC
[asterisk-users] getting real sip status after dial
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(${CHANNEL(name)},tech)})
>
> Remember the incoming leg of the call and the outgoing leg of the call
> are different channels. Make sure you are giving HANGUPCAUSE the
> correct channel.
>
> On 06/09/2018 02:01 PM, Khalil Khamlichi wrote:
> > It seems very weird to me that we cannot access sip code of a call
> > from pjsip which information is actually returned from the provider,
> > so it is available to asterisk, why does asterisk hide it ?
> > On Sat, Jun 9, 2018 at 5:08 PM Khalil Khamlichi
> > <khamlichi.khalil at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >> Is there any way I can get exact sip status from pjsip after a
dial ?
> >> or all we can
> >> get is asterisk hangup causes ?
> >>
> >> Thanks in advance.
> >>
> >> KKh
> >
>
> --
> http://help.nyigc.net/
My actual dialplan hangupcause handling is in AGI and AEL, so the
dialplan below has not been tested.
Don't use HANGUPCAUSE_KEYS, the order of the channels varies. Save the
channel name while the dialplan is in the predial handler like below.
MASTER_CHANNEL is used to avoid silly issues with the scope of the
out_chan variable.
[test]
exten => _X.,1,Set(CHANNEL(hangup_handler_push)=caller_hangup,s,1)
same => n,Dial(SIP/my-peer/12125551212,30,b(test_pre_dial^s^1))
[test_pre_dial]
; while in the handler save the channel
exten => _X.,1,Set(MASTER_CHANNEL(out_chan)=${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/
--
http://help.nyigc.net/