[OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite

Sebastian Sastre sastre.sebastian at gmail.com
Thu Aug 2 09:27:39 EDT 2018


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.sebastian at 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 <razvan at 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.sebastian at gmail.com <mailto:sastre.sebastian at 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 at 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:user at domain.com <mailto:
>>> sip%3Auser at domain.com>
>>>              to_uri:: sip:18889990000 at domain.com
>>>     <mailto:sip%3A18889990000 at domain.com>
>>>              caller_tag:: 1i4vfmjico
>>>              caller_contact:: sip:lccpphv2 at 192.168.202.3
>>>     <mailto:sip%3Alccpphv2 at 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 at 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:user at domain.com <mailto:
>>> sip%3Auser at domain.com>
>>>              to_uri:: sip:18889990000 at domain.com
>>>     <mailto:sip%3A18889990000 at domain.com>
>>>              caller_tag:: 1i4vfmjico
>>>              caller_contact:: sip:lccpphv2 at 192.168.202.3
>>>     <mailto:sip%3Alccpphv2 at 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 <razvan at opensips.org
>>>     <mailto:razvan at 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 !
>>>
>>>
>>>
>>>             _______________________________________________
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>>>             <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
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>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>         <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>
>>>
>>>
>>>
>>>
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>>>
>> --
>> Răzvan Crainea
>> OpenSIPS Core Developer
>>   http://www.opensips-solutions.com
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
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