On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens
<jonas.kell...@telenet.be <mailto:jonas.kell...@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@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@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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users