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

[Răzvan Crainea]

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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> Users at lists.opensips.org
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> 

-- 
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com