[asterisk-bugs] Possible SIP bug, needs confirmation/testing

[Robert Dyck]

I may have found an asterisk bug but I do not have access to an asterisk site 
for testing. I would be happy to work with someone to confirm this possible 
bug.

Unfortunately my service provider has ignored my reports and has not even told 
me which version they are using. Perhaps someone in the community with more 
influence or with insider knowledge could determine which version 
voxalot/sipbroker is using.

The scenario is a call to an asterisk PSTN gateway where the gateway performs 
a re-INVITE. Calls in the other direction seem to work well. The specific 
problem is that the UAS times out waiting for the ACK after the re-INVITE. 
The complicating factor is that the UAS does not send an Record-Route list 
with its in-dialogue 200 OK. This does not violate the spec which says the 
UAS MAY send RR. The spec also says the RR list should have no influence on 
the existing route set and by implication the lack of an RR list should not 
create an empty route set. I suspect that asterisk is creating an empty route 
set under these conditions and that is why the ACK does not reach the UAS. 
Note that when the UAS times out it sends BYE this time including an RR list 
and the 200 OK for the BYE reaches the UA.

A route set that can be altered in dialogue is probably a security threat.

Thank you, Rob