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

[Sebastian Sastre]

​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
        to_uri:: sip:18889990000 at domain.com
        caller_tag:: 1i4vfmjico
        caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob
        callee_cseq:: 0
        caller_route_set::
        caller_bind_addr:: wss:10.101.10.151:443
        caller_sdp::
        CALLEES::
                callee::
                        callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee
                        callee_contact:: sip:10.101.10.161:5060
                        caller_cseq:: 1094
                        callee_route_set::
                        callee_bind_addr:: udp: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
        to_uri:: sip:18889990000 at domain.com
        caller_tag:: 1i4vfmjico
        caller_contact:: sip:lccpphv2 at 192.168.202.3:51292;transport=wss;ob
        callee_cseq:: 0
        caller_route_set::
        caller_bind_addr:: wss:10.101.10.151:443
        caller_sdp::
        CALLEES::
                callee::
                        callee_tag:: d651df12-c9c2-4db1-99ad-b15d6240ffee
                        callee_contact:: sip:10.101.10.161:5060
                        caller_cseq:: 1095
                        callee_route_set::
                        callee_bind_addr:: udp:10.101.10.151:5060
                        callee_sdp::


On Mon, Jul 16, 2018 at 7:55 AM, Răzvan Crainea <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 !
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at 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 at lists.opensips.org
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>
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