<div dir="ltr"><div class="gmail_default" style="font-family:tahoma,sans-serif">I put a full debug on this paste <a href="https://pastebin.com/BEJ6fAR8">https://pastebin.com/BEJ6fAR8</a> </div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif">Should be </div><div class="gmail_default" style="font-family:tahoma,sans-serif"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;white-space:pre-wrap"> Jul 13 11:42:39 gcwregistrar151 /sbin/opensips[3647]: ERROR:tm:msg_send: send() to <a href="http://192.0.2.246:443">192.0.2.246:443</a> for proto wss/6 failed</span><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif"><span style="color:rgb(0,0,0);font-family:arial,sans-serif;white-space:pre-wrap"><br></span></div><div class="gmail_default"><span style="orphans: 2; widows: 2;"><font color="#000000"><span style="white-space:pre-wrap">Thanks</span></font> </span></div><div class="gmail_default"><span style="orphans: 2; widows: 2;"><font color="#000000"><span style="white-space:pre-wrap"><br></span></font></span></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Jul 23, 2018 at 11:47 AM, Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi, Sebastien!<br>
<br>
It looks like the contact is not changing. Can you indicate what connection he is trying to find, i.e. what IP, port and proto? You should see them in the error line.<br>
<br>
Best regards,<br>
Răzvan<span class=""><br>
<br>
On 07/23/2018 06:43 PM, Sebastian Sastre wrote:<br>
</span><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class="">
Hey Razvan,<br>
<br>
I’ve been playing around a lot with this but I can’t seem to make it work. Whatever I do without the fix route doesn’t find a suitable tcp connection.<br>
<br>
Do you see something on the dlg_list before that would indicate the problem Or is there any other debug I can use ?<br>
<br>
Thanks again !<br>
<br>
<br></span><span class="">
On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre <<a href="mailto:sastre.sebastian@gmail.com" target="_blank">sastre.sebastian@gmail.com</a> <mailto:<a href="mailto:sastre.sebastian@gmail.com" target="_blank">sastre.sebastian@gmail<wbr>.com</a>>> wrote:<br>
<br>
Razvan,<br>
Thanks ! I tried what you indicated but I don’t see the contact<br>
changing. Im taking care of the fix contacts where it needs to be<br>
bet but still on the bye it can’t find it.<br>
<br>
<br></span>
root@gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)*<span class=""><br>
dialog:: ID=5820137817639<br>
state:: 4<br>
user_flags:: 0<br>
timestart:: 1532043804<br>
datestart:: 2018-07-19 19:43:24<br>
timeout:: 1532044163<br>
dateout:: 2018-07-19 19:49:23<br>
callid:: fp436dll6pcmdqk78gn6<br></span>
from_uri:: <a href="mailto:sip%3Auser@domain.com" target="_blank">sip:user@domain.com</a> <mailto:<a href="mailto:sip%253Auser@domain.com" target="_blank">sip%3Auser@domain.com</a>><br>
to_uri:: <a href="mailto:sip%3A18889990000@domain.com" target="_blank">sip:18889990000@domain.com</a><br>
<mailto:<a href="mailto:sip%253A18889990000@domain.com" target="_blank">sip%3A18889990000@doma<wbr>in.com</a>><span class=""><br>
caller_tag:: 1i4vfmjico<br>
caller_contact:: <a href="mailto:sip%3Alccpphv2@192.168.202.3" target="_blank">sip:lccpphv2@192.168.202.3</a><br></span>
<mailto:<a href="mailto:sip%253Alccpphv2@192.168.202.3" target="_blank">sip%3Alccpphv2@192.168<wbr>.202.3</a>>:51292;transport=wss;ob<span class=""><br>
callee_cseq:: 0<br>
caller_route_set::<br>
caller_bind_addr:: wss:<a href="http://10.101.10.151:443" rel="noreferrer" target="_blank">10.101.10.151:443</a><br></span>
<<a href="http://10.101.10.151:443" rel="noreferrer" target="_blank">http://10.101.10.151:443</a>><span class=""><br>
caller_sdp::<br>
CALLEES::<br>
callee::<br>
callee_tag::<br>
d651df12-c9c2-4db1-99ad-b15d62<wbr>40ffee<br>
callee_contact:: sip:<a href="http://10.101.10.161:5060" rel="noreferrer" target="_blank">10.101.10.161:5060</a><br></span>
<<a href="http://10.101.10.161:5060" rel="noreferrer" target="_blank">http://10.101.10.161:5060</a>><span class=""><br>
caller_cseq:: 1094<br>
callee_route_set::<br>
callee_bind_addr:: udp:<a href="http://10.101.10.151:5060" rel="noreferrer" target="_blank">10.101.10.151:5060</a><br></span>
<<a href="http://10.101.10.151:5060" rel="noreferrer" target="_blank">http://10.101.10.151:5060</a>><br>
callee_sdp::<br>
<br>
root@gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )*<span class=""><br>
dialog:: ID=5820137817639<br>
state:: 4<br>
user_flags:: 0<br>
timestart:: 1532043804<br>
datestart:: 2018-07-19 19:43:24<br>
timeout:: 1532044163<br>
dateout:: 2018-07-19 19:49:23<br>
callid:: fp436dll6pcmdqk78gn6<br></span>
from_uri:: <a href="mailto:sip%3Auser@domain.com" target="_blank">sip:user@domain.com</a> <mailto:<a href="mailto:sip%253Auser@domain.com" target="_blank">sip%3Auser@domain.com</a>><br>
to_uri:: <a href="mailto:sip%3A18889990000@domain.com" target="_blank">sip:18889990000@domain.com</a><br>
<mailto:<a href="mailto:sip%253A18889990000@domain.com" target="_blank">sip%3A18889990000@doma<wbr>in.com</a>><span class=""><br>
caller_tag:: 1i4vfmjico<br>
caller_contact:: <a href="mailto:sip%3Alccpphv2@192.168.202.3" target="_blank">sip:lccpphv2@192.168.202.3</a><br></span>
<mailto:<a href="mailto:sip%253Alccpphv2@192.168.202.3" target="_blank">sip%3Alccpphv2@192.168<wbr>.202.3</a>>:51292;transport=wss;ob<span class=""><br>
callee_cseq:: 0<br>
caller_route_set::<br>
caller_bind_addr:: wss:<a href="http://10.101.10.151:443" rel="noreferrer" target="_blank">10.101.10.151:443</a><br></span>
<<a href="http://10.101.10.151:443" rel="noreferrer" target="_blank">http://10.101.10.151:443</a>><span class=""><br>
caller_sdp::<br>
CALLEES::<br>
callee::<br>
callee_tag::<br>
d651df12-c9c2-4db1-99ad-b15d62<wbr>40ffee<br>
callee_contact:: sip:<a href="http://10.101.10.161:5060" rel="noreferrer" target="_blank">10.101.10.161:5060</a><br></span>
<<a href="http://10.101.10.161:5060" rel="noreferrer" target="_blank">http://10.101.10.161:5060</a>><span class=""><br>
caller_cseq:: 1095<br>
callee_route_set::<br>
callee_bind_addr:: udp:<a href="http://10.101.10.151:5060" rel="noreferrer" target="_blank">10.101.10.151:5060</a><br></span>
<<a href="http://10.101.10.151:5060" rel="noreferrer" target="_blank">http://10.101.10.151:5060</a>><span class=""><br>
callee_sdp::<br>
<br>
<br>
On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea <<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a><br></span><div><div class="h5">
<mailto:<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>>> wrote:<br>
<br>
Hi, Sebastian!<br>
<br>
The re-invite probably generates a remote contact update. And if<br>
you don't "fix" the contact on re-invites and their 200 OK, you<br>
might end up with broken contacts in the dialog, thus sequential<br>
signaling will not work.<br>
I suggest you do two things to debug this:<br>
1. remove the fix_route_dialog() call - the call should still be<br>
routed according to RR information, presuming this information<br>
is correct.<br>
2. start the call, run `opensipsctl fifo dlg_list` and write<br>
down the WSS's contact, then put the call on hold, and check<br>
again the contact.<br>
<br>
Best regards,<br>
Răzvan<br>
<br>
<br>
On 07/13/2018 09:19 PM, Sebastian Sastre wrote:<br>
<br>
<br>
Hello, I’ve been experiencing a situation with Proto WSS.<br>
The scenario is very simple. A call is established from an<br>
Asterisk Box to Opensips (UDP) and finally a SipJs7.8 (WSS).<br>
Everything works great and we are able to register using mid<br>
registrar and pass calls thru.<br>
<br>
When an agent puts the call on hold a reinvite is correctly<br>
negotiated and the call is placed on hold and viceversa. However!, if the originating caller disconnects the call<br>
while still on hold, Asterisk will correctly terminate the<br>
dialog with a Bye but when OpenSIPs will complain about not<br>
finding a suitable tcp connection and responds with a 477<br>
even after successfully matching and processing the dialog<br>
termination correctly.<br>
<br>
opensipsctl fifo list_tcp_conns shows the connection available.<br>
<br>
The only way I found of fixing this problem is by adding<br>
fix_route_dialog() on the sequential loose route.<br>
<br>
if (loose_route()) {<br>
if (is_method("BYE")) {<br>
if (!validate_dialog()){<br>
fix_route_dialog();<br>
}<br>
<br>
What do you guys think?<br>
Am I messing up something in the script or is this the<br>
correct way to address this problem?<br>
<br>
The funny thing is that there is no difference notable<br>
between the bye after hold and a regular bye without putting<br>
the call on hold.<br>
Here is the opensips log with the error and the trace.<br>
<br>
<a href="https://pastebin.com/BEJ6fAR8" rel="noreferrer" target="_blank">https://pastebin.com/BEJ6fAR8</a><br>
<br>
Thanks !<br>
<br>
<br>
<br>
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<br>
<br>
-- Răzvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" rel="noreferrer" target="_blank">http://www.opensips-solutions.<wbr>com</a><br>
<<a href="http://www.opensips-solutions.com" rel="noreferrer" target="_blank">http://www.opensips-solutions<wbr>.com</a>><br>
<br>
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<br>
<br>
<br>
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<br>
</span></blockquote><div class="HOEnZb"><div class="h5">
<br>
-- <br>
Răzvan Crainea<br>
OpenSIPS Core Developer<br>
<a href="http://www.opensips-solutions.com" rel="noreferrer" target="_blank">http://www.opensips-solutions.<wbr>com</a><br>
<br>
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