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<blockquote type="cite"
cite="mid:CAM0A2Z0j=JvN_wAoDWxjavKUXYcPu4-5AtPSwMd+puF4Wwtaug@mail.gmail.com">
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<div dir="ltr">On Fri, Feb 11, 2022 at 9:31 AM Jonas Kellens
<<a href="mailto:jonas.kellens@telenet.be"
moz-do-not-send="true">jonas.kellens@telenet.be</a>>
wrote:<br>
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<p><font face="Helvetica, Arial, sans-serif">Hello</font></p>
<p><font face="Helvetica, Arial, sans-serif"><br>
</font></p>
<p><font face="Helvetica, Arial, sans-serif">I notice a
major difference in what Asterisk console is telling
me (which seems correct) and what Asterisk Manager is
telling.</font></p>
<p><br>
</p>
<p>A SIP user is called, and the phone does not ring. This
is the situation.</p>
<p><br>
</p>
<p>On Asterisk console I see (which seems to be in line
with an unreachable phone) :</p>
<p>[Feb 11 11:31:31] VERBOSE[15653][C-00000319]
app_dial.c: Called SIP/mysipuser6<br>
[Feb 11 11:31:37] VERBOSE[15653][C-00000319] app_dial.c:
Everyone is busy/congested at this time (1:0/0/1)<br>
[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</p>
<p>However on Asterisk Manager interface I see the event :</p>
<p>11:31:31<br>
Array<br>
(<br>
[0] => Event: DeviceStateChange<br>
[1] => Privilege: call,all<br>
[2] => SystemName: voipserver1<br>
[3] => Device: SIP/mysipuser6<br>
[4] => State: RINGING<br>
)</p>
<p><br>
</p>
<p>I can reproduce this easily every time :</p>
<p>[Feb 11 11:31:46] VERBOSE[15719][C-0000031a]
app_dial.c: Called SIP/mysipuser6<br>
[Feb 11 11:31:53] VERBOSE[15719][C-0000031a] app_dial.c:
Everyone is busy/congested at this time (1:0/0/1)<br>
[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</p>
<p>11:31:46<br>
Array<br>
(<br>
[0] => Event: DeviceStateChange<br>
[1] => Privilege: call,all<br>
[2] => SystemName: voipserver1<br>
[3] => Device: SIP/mysipuser6<br>
[4] => State: RINGING<br>
)</p>
<p><br>
</p>
<p>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).<br>
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<div>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.</div>
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-- <br>
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<div style="font-family:tahoma,sans-serif"><font
color="#073763">Joshua C. Colp</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Asterisk Technical Lead</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Sangoma Technologies</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Check us out at <a
href="http://www.sangoma.com"
target="_blank" moz-do-not-send="true">www.sangoma.com</a>
and <a href="http://www.asterisk.org"
target="_blank" moz-do-not-send="true">www.asterisk.org</a></font><br>
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<br>
<br>
So if "DeviceStateChange" is not reporting the real state of a SIP
user/device (like 180-ringing), which event does ?!<br>
<br>
<br>
<br>
Kind regards.<br>
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