Juan Carlos Castro y Castro wrote:
I have a scenario in which an OpenSER machine
distributes load among
several Asterisk machines for outgoing PSTN. I use an external program
which I call through exec_dset() to select which server a call goes to.
I use record_route().
All calls are completed OK, but when I dial from some SIP user-agents
and hang up from the caller side, OpenSER gets crazy and doesn't know
where to route the BYE to. Here's the reason:
Good scenario:
- I make a call to <5551234@mydomain>
- OpenSER calls exec_dset, which turns it into
<providercode#5551234@chosenserver>
- Call is answered
- I hang up
- My UA sends a BYE like this: "BYE
sip:providercode#5551234@chosenserver SIP/2.0"
- I get an OK back, all is well.
(Sometimes the # in the uri above is sent as %23, but it works either
way)
Bad scenario:
- I make a call to <5551234@mydomain>
- OpenSER calls exec_dset, which turns it into
<providercode#5551234@chosenserver>
- Call is answered
- I hang up
- My UA sends a BYE like this: "BYE sip:5551234@mydomain SIP/2.0"
- My openser.cfg doesn't know how to handle that and I get a "Loop
Detected" back. The destination never gets a hangup signal.
Is the second UA from the "bad" scenario disrespecting the RFC? Do I
have any recourse to route a BYE that comes like that to the right
Asterisk server?
Hi, Juan!
If called UA use 'providercode#5551234@chosenserver' as it own
Contact-header then calling UA must use it to construct R-URI for BYE
request.
But I think that called party still use '5551234@mydomain' as
Contact-header. So you must route BYE request by the same way as INVITE
request.
--
CU,
Victor Gamov