I put a full debug on this paste https://pastebin.com/BEJ6fAR8
Should be Jul 13 11:42:39 gcwregistrar151 /sbin/opensips[3647]: ERROR:tm:msg_send: send() to 192.0.2.246:443 for proto wss/6 failed Thanks On Mon, Jul 23, 2018 at 11:47 AM, Răzvan Crainea <[email protected]> wrote: > Hi, Sebastien! > > 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. > > Best regards, > Răzvan > > On 07/23/2018 06:43 PM, Sebastian Sastre wrote: > >> Hey Razvan, >> >> 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. >> >> Do you see something on the dlg_list before that would indicate the >> problem Or is there any other debug I can use ? >> >> Thanks again ! >> >> >> On Thu, Jul 19, 2018 at 8:03 PM, Sebastian Sastre < >> [email protected] <mailto:[email protected]>> wrote: >> >> Razvan, >> Thanks ! I tried what you indicated but I don’t see the contact >> changing. Im taking care of the fix contacts where it needs to be >> bet but still on the bye it can’t find it. >> >> >> root@gcwregistrar151:~$ opensipsctl fifo dlg_list. *(Call Connected)* >> dialog:: ID=5820137817639 >> state:: 4 >> user_flags:: 0 >> timestart:: 1532043804 >> datestart:: 2018-07-19 19:43:24 >> timeout:: 1532044163 >> dateout:: 2018-07-19 19:49:23 >> callid:: fp436dll6pcmdqk78gn6 >> from_uri:: sip:[email protected] <mailto:sip%[email protected] >> > >> to_uri:: sip:[email protected] >> <mailto:sip%[email protected]> >> caller_tag:: 1i4vfmjico >> caller_contact:: sip:[email protected] >> <mailto:sip%[email protected]>:51292;transport=wss;ob >> callee_cseq:: 0 >> caller_route_set:: >> caller_bind_addr:: wss:10.101.10.151:443 >> <http://10.101.10.151:443> >> caller_sdp:: >> CALLEES:: >> callee:: >> callee_tag:: >> d651df12-c9c2-4db1-99ad-b15d6240ffee >> callee_contact:: sip:10.101.10.161:5060 >> <http://10.101.10.161:5060> >> caller_cseq:: 1094 >> callee_route_set:: >> callee_bind_addr:: udp:10.101.10.151:5060 >> <http://10.101.10.151:5060> >> callee_sdp:: >> >> root@gcwregistrar151:~$ opensipsctl fifo dlg_list *(Call on Hold )* >> dialog:: ID=5820137817639 >> state:: 4 >> user_flags:: 0 >> timestart:: 1532043804 >> datestart:: 2018-07-19 19:43:24 >> timeout:: 1532044163 >> dateout:: 2018-07-19 19:49:23 >> callid:: fp436dll6pcmdqk78gn6 >> from_uri:: sip:[email protected] <mailto:sip%[email protected] >> > >> to_uri:: sip:[email protected] >> <mailto:sip%[email protected]> >> caller_tag:: 1i4vfmjico >> caller_contact:: sip:[email protected] >> <mailto:sip%[email protected]>:51292;transport=wss;ob >> callee_cseq:: 0 >> caller_route_set:: >> caller_bind_addr:: wss:10.101.10.151:443 >> <http://10.101.10.151:443> >> caller_sdp:: >> CALLEES:: >> callee:: >> callee_tag:: >> d651df12-c9c2-4db1-99ad-b15d6240ffee >> callee_contact:: sip:10.101.10.161:5060 >> <http://10.101.10.161:5060> >> caller_cseq:: 1095 >> callee_route_set:: >> callee_bind_addr:: udp:10.101.10.151:5060 >> <http://10.101.10.151:5060> >> callee_sdp:: >> >> >> On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea <[email protected] >> <mailto:[email protected]>> wrote: >> >> Hi, Sebastian! >> >> The re-invite probably generates a remote contact update. And if >> you don't "fix" the contact on re-invites and their 200 OK, you >> might end up with broken contacts in the dialog, thus sequential >> signaling will not work. >> I suggest you do two things to debug this: >> 1. remove the fix_route_dialog() call - the call should still be >> routed according to RR information, presuming this information >> is correct. >> 2. start the call, run `opensipsctl fifo dlg_list` and write >> down the WSS's contact, then put the call on hold, and check >> again the contact. >> >> Best regards, >> Răzvan >> >> >> On 07/13/2018 09:19 PM, Sebastian Sastre wrote: >> >> >> Hello, I’ve been experiencing a situation with Proto WSS. >> The scenario is very simple. A call is established from an >> Asterisk Box to Opensips (UDP) and finally a SipJs7.8 (WSS). >> Everything works great and we are able to register using mid >> registrar and pass calls thru. >> >> When an agent puts the call on hold a reinvite is correctly >> negotiated and the call is placed on hold and viceversa. >> However!, if the originating caller disconnects the call >> while still on hold, Asterisk will correctly terminate the >> dialog with a Bye but when OpenSIPs will complain about not >> finding a suitable tcp connection and responds with a 477 >> even after successfully matching and processing the dialog >> termination correctly. >> >> opensipsctl fifo list_tcp_conns shows the connection >> available. >> >> The only way I found of fixing this problem is by adding >> fix_route_dialog() on the sequential loose route. >> >> if (loose_route()) { >> if (is_method("BYE")) { >> if (!validate_dialog()){ >> fix_route_dialog(); >> } >> >> What do you guys think? >> Am I messing up something in the script or is this the >> correct way to address this problem? >> >> The funny thing is that there is no difference notable >> between the bye after hold and a regular bye without putting >> the call on hold. >> Here is the opensips log with the error and the trace. >> >> https://pastebin.com/BEJ6fAR8 >> >> Thanks ! >> >> >> >> _______________________________________________ >> Users mailing list >> [email protected] <mailto:[email protected]> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >> >> >> -- Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> <http://www.opensips-solutions.com> >> >> _______________________________________________ >> Users mailing list >> [email protected] <mailto:[email protected]> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >> >> >> >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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