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<div>Went through a git bisect on this, and tracked the behavior
change to this commit;</div>
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<div>commit 40bfe0fff566476ff95242581f391f5b5b1c7e32<br>
Author: Anthony Minessale <a class="moz-txt-link-rfc2396E" href="mailto:anthm@freeswitch.org"><anthm@freeswitch.org></a><br>
Date: Mon Jan 16 15:27:36 2017 -0600<br>
<br>
FS-9154: [freeswitch-core] Add & remove video on
re-invites #resolve</div>
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<div>Note that these calls do NOT have any video. <br>
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<div>I've added this info to FS-9983 jira issue which looks highly
related, but will go ahead and file a new jira issue as well. <br>
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<div>-- Nathan<br>
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<div class="moz-cite-prefix"><span style="margin: -1.3px 0 0 0
!important;"><font style="font: 13px Tahoma !important; color:
#000000 !important;" face="Tahoma" color="#000000"><b><br>
</b></font></span></div>
<div class="moz-cite-prefix"><span style="margin: -1.3px 0 0 0
!important;"><font style="font: 13px Tahoma !important; color:
#000000 !important;" face="Tahoma" color="#000000"><b>From:</b>
Nathan Neulinger <a class="moz-txt-link-rfc2396E" href="mailto:nneul@mst.edu"><nneul@mst.edu></a></font></span><br>
<div><span style="margin: -1.3px 0 0 0 !important;"><font
style="font: 13px Tahoma !important; color: #000000
!important;" face="Tahoma" color="#000000"><b>Sent:</b> Thu,
Sep 6, 2018 10:48 PM CDT</font></span><br>
<span style="margin: -1.3px 0 0 0 !important;"><font
style="font: 13px Tahoma !important; color: #000000
!important;" face="Tahoma" color="#000000"><b>To:</b>
FreeSWITCH Users Help
<a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a></font></span><br>
<span style="margin: -1.3px 0 0 0 !important;"><font
style="font: 13px Tahoma !important; color: #000000
!important;" face="Tahoma" color="#000000"><b>Subject:</b>
Phone spontaneously going into Hold when a call is
established?</font></span><br>
<br>
</div>
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<blockquote type="cite"
cite="mid:c7b4f0f5-c953-f49a-5172-f4754f9fd905@mst.edu"
style="border:none !important; margin-left:0px !important;
margin-right:0px !important; margin-top:0px !important;
padding-left:0px !important; padding-right:0px !important">Finally
managed to get a reproducible failure on this, but don't
understand SIP enough to know where it's going wrong. I have two
local numbers that I can call via my local telco over a SIP trunk
(MetaSwitch) - one reliably demonstrates the behavior, the other
reliably works normal:
<br>
<br>
The call that is working normally (leg w/ provider):
<br>
INVITE w/ neither sendonly nor sendrecv
<br>
100 Trying
<br>
183 Progress w/ a=sendrecv
<br>
200 OK w/ a=sendrecv
<br>
<br>
The call that is going into on-hold immediately (leg w/ provider):
<br>
INVITE w/ neither sendonly nor sendrecv
<br>
100 Trying
<br>
180 Ringing
<br>
200 OK w/ a=sendonly
<br>
INVITE w/ a=sendrecv
<br>
100 Trying
<br>
200 OK w/ neither sendonly nor sendrecv
<br>
<br>
Call that goes into on hold immediately (leg w/ device):
<br>
INVITE w/ neither, but has sdp
<br>
407
<br>
INVITE w/ neither, but has sdp
<br>
100
<br>
183 w/ neither
<br>
200 w/ neither
<br>
UPDATE w/ sendonly (which is after the second 200 OK above)
<br>
200 OK w/ recvonly
<br>
<br>
<br>
<br>
I'm going to try and do some more digging on this tomorrow, but if
someone has any insights here on what the proper/expected behavior
is, I sure would love to hear it.
<br>
<br>
It seems like this may be somewhat related to FS-9765 - however
there is no invite w/o an SDP. Also similar sounding to FS-11234.
I'm happy to take this forward to provider (I'm inquiring with
them as well) but would be helpful to have some more details on
which piece is behaving incorrectly.
<br>
<br>
-- Nathan
<br>
<br>
-------- Original Message --------
<br>
From: Nathan Neulinger
<br>
Sent: Tue, Sep 4, 2018 2:45 PM CDT
<br>
To: FreeSWITCH Users Help
<a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a>
<br>
Subject: Phone spontaneously going into Hold when a call is
established?
<br>
<br>
I had seen this symptom a few times randomly, and just got a
couple of reports from users where a call to an outside
<br>
number was going immediately to 'On Hold' when the person
answered.
<br>
<br>
Is there anything in the SIP/RTP stream that could be expected to
trigger that sort of behavior? I've seen it from a
<br>
polycom, and I believe one of the user reports today was with a
yealink. User report was with a recently upgraded 1.6 ->
<br>
1.8 deployment. I believe the previous times I've seen it would
have been on git builds of 1.8, but not certain.
<br>
<br>
-- Nathan
<br>
------------------------------------------------------------
<br>
Nathan <a class="moz-txt-link-abbreviated" href="mailto:Neulingernneul@mst.edu">Neulingernneul@mst.edu</a>
<br>
Missouri S&T Information Technology (573) 612-1412
<br>
System Administrator - Architect
<br>
<br>
<br>
</blockquote>
<pre class="moz-signature" cols="120">--
------------------------------------------------------------
Nathan Neulinger <a class="moz-txt-link-abbreviated" href="mailto:nneul@mst.edu">nneul@mst.edu</a>
Missouri S&T Information Technology (573) 612-1412
System Administrator - Architect </pre>
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