[OpenSIPS-Users] Proto WSS No Open TCP Connection after reinvite
Sebastian Sastre
sastre.sebastian at gmail.com
Mon Jul 23 11:58:02 EDT 2018
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 !
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org <mailto:Users at 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 at lists.opensips.org <mailto:Users at 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 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
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20180723/4a165f6b/attachment-0001.html>
More information about the Users
mailing list