Hey guys, sorry to keep insisting, im just very curious why its not finding that TCP connection.
Nothing anywhere tells or yields an error. What else can I look at? On Mon, Jul 23, 2018 at 11:58 AM, Sebastian Sastre < sastre.sebast...@gmail.com> wrote: > 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 <raz...@opensips.org> > 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 < >>> sastre.sebast...@gmail.com <mailto:sastre.sebast...@gmail.com>> 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:u...@domain.com <mailto: >>> sip%3au...@domain.com> >>> to_uri:: sip:18889990...@domain.com >>> <mailto:sip%3a18889990...@domain.com> >>> caller_tag:: 1i4vfmjico >>> caller_contact:: sip:lccpphv2@192.168.202.3 >>> <mailto:sip%3Alccpphv2@192.168.202.3>: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:u...@domain.com <mailto: >>> sip%3au...@domain.com> >>> to_uri:: sip:18889990...@domain.com >>> <mailto:sip%3a18889990...@domain.com> >>> caller_tag:: 1i4vfmjico >>> caller_contact:: sip:lccpphv2@192.168.202.3 >>> <mailto:sip%3Alccpphv2@192.168.202.3>: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 <raz...@opensips.org >>> <mailto:raz...@opensips.org>> 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 >>> Users@lists.opensips.org <mailto:Users@lists.opensips.org> >>> 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 >>> Users@lists.opensips.org <mailto:Users@lists.opensips.org> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users> >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> -- >> Răzvan Crainea >> OpenSIPS Core Developer >> http://www.opensips-solutions.com >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >
_______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users