[asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange

[Jonas Kellens]

> On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens 
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
>     Hello
>
>
>     I notice a major difference in what Asterisk console is telling me
>     (which seems correct) and what Asterisk Manager is telling.
>
>
>     A SIP user is called, and the phone does not ring. This is the
>     situation.
>
>
>     On Asterisk console I see (which seems to be in line with an
>     unreachable phone) :
>
>     [Feb 11 11:31:31] VERBOSE[15653][C-00000319] app_dial.c: Called
>     SIP/mysipuser6
>     [Feb 11 11:31:37] VERBOSE[15653][C-00000319] app_dial.c: Everyone
>     is busy/congested at this time (1:0/0/1)
>     [Feb 11 11:31:37] VERBOSE[15653][C-00000319] pbx.c: Executing
>     [202 at from-PBX:253] NoOp("SIP/mysipuser12-0000157d",
>     "DIALSTATUS=CHANUNAVAIL") in new stack
>
>     However on Asterisk Manager interface I see the event :
>
>     11:31:31
>     Array
>     (
>         [0] => Event: DeviceStateChange
>         [1] => Privilege: call,all
>         [2] => SystemName: voipserver1
>         [3] => Device: SIP/mysipuser6
>         [4] => State: RINGING
>     )
>
>
>     I can reproduce this easily every time :
>
>     [Feb 11 11:31:46] VERBOSE[15719][C-0000031a] app_dial.c: Called
>     SIP/mysipuser6
>     [Feb 11 11:31:53] VERBOSE[15719][C-0000031a] app_dial.c: Everyone
>     is busy/congested at this time (1:0/0/1)
>     [Feb 11 11:31:53] VERBOSE[15719][C-0000031a] pbx.c: Executing
>     [202 at from-PBX:253] NoOp("SIP/mysipuser12-0000157f",
>     "DIALSTATUS=CHANUNAVAIL") in new stack
>
>     11:31:46
>     Array
>     (
>         [0] => Event: DeviceStateChange
>         [1] => Privilege: call,all
>         [2] => SystemName: voipserver1
>         [3] => Device: SIP/mysipuser6
>         [4] => State: RINGING
>     )
>
>
>     Why is Asterisk Manager reporting a RINGING state if there is no
>     SIP 180 RINGING received ?! When issuing a SIP DEBUG, I see a SIP
>     INVITE but no response (so no SIP 180 or 183).
>
>
> The answer seems to be, because that's the way chan_sip was written. 
> As soon as an outgoing call is attempted it sets some internal state 
> to ringing, which is then used when it reports device state 
> information. DeviceStateChange is just reporting what chan_sip told it.
>
> -- 
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com <http://www.sangoma.com> and 
> www.asterisk.org <http://www.asterisk.org>


So if "DeviceStateChange" is not reporting the real state of a SIP 
user/device (like 180-ringing), which event does ?!



Kind regards.
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