Joshua C. Colp
2022-Feb-11 13:37 UTC
[asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange
On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens <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 and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220211/d63a0582/attachment.html>
Jonas Kellens
2022-Feb-11 13:44 UTC
[asterisk-users] Discrepancy between Asterisk console and Asterisk Manager DeviceStateChange
> 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220211/6b34a71d/attachment.html>