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