Route headers are fine - the problem is the RURI of the BYE:
See the Contact header of the INVITE:
Contact: <sip:[EMAIL PROTECTED]:4294;transport=TLS>
This URI must be used in the RURI of the BYE, but Asterisk uses:
BYE sip:[EMAIL PROTECTED]:4294 SIP/2.0
Thus, the proxy forwards the request with UDP instead of TLS. Thus, this
is a bug in Asterisk. Try update Asterisk. Try looking at Asterisk Bug
tracker for this bug. If you are unlucky, open a bug report on the
Asterisk bug tracker (bugs.digium.com)
regards
klaus
David Loh schrieb:
Hi,
Arrggghh .. that's one of my attempts to eliminate the broken "BYE"
problem... that's ngrep was captured when I set "modparam("rr",
"enable_double_rr", "0");",
I've paste another ngrep to http://pastebin.ca/674450, this time the
double RR header is enabled.
And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt
the post is "openser").
Even though double RR header is enabled, but for BYE it's still doesn't
process properly :(
For the .cfg file line #130 onward, I did tried t_relay, forward and
force_send_socket,
but none of this will do the trick (force_send_socket was complaining
TLS error due to missing certificate (?) )
Would appreciate if anyone could enlighten me why is this happen ?
Thanks,
David Loh
Klaus Darilion wrote:
But the INVITE you posted at http://pastebin.ca/673392 also has only
one Record-Route header.
regards
klaus
David Loh schrieb:
Hi,
Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK
...but when asterisk disconnected the call and send BYE back to OpenSER,
the TLS RR header wasn't present, the only 2 RR header was
"SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" ....
I'm puzzled ... is there any command to 'fix' this?
Regards,
David Loh
Klaus Darilion wrote:
The openser proxy should add 2 record-route header (TLS and UDP =
double record route). This is why it does not work.
regards
klaus
David Loh schrieb:
Hi All,
Greeting.
I've been struggle with OpenSER TLS implementation for more than a
week, since I've ported from UDP to TLS, everything work fine
except the "BYE" request from Asterisk (loose route), my
implementation was something like below:
[Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk]
My OpenSER.cfg already configured to listen on two port which is :-
"tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or
even voicemail) having no problem,
but when the callee disconnect the call, caller will never get hang
up :(
I've attached my ethereal trace/ngrep to pastebin,
http://pastebin.ca/673392
Wondering if anyone can help me with the broken "BYE" that returned
from Asterisk ?
Line #131, supposedly this line should have contain 2 Via header,
one was "SIP/2.0/UDP" and another "SIP/2.0/TLS",
but somehow the TLS via header was gone !! (compare to previous ACK
(Line #117) /INVITE (Line #51).
Due to the missing TLS via header, OpenSER log file was complaining
"protocol/port mis-match".
The last BYE request (Line #256) is actually firing from Client,
which contain the "TLS" via.
I've even tried "force_send_socket" to port 5061 (instead of 5060)
from loose route, but it complaining TLS certificate error,
since Asterisk doesn't support TLS natively, I've no clue why is
the ACK/INVITE/CANCEL work but not BYE.
if (loose_route) {
....
if(is_method("BYE")) { force_send_socket(IP:5061); }
}
Has any one gone through of this kinda OpenSER over TLS + Asterisk
setup,
I'm really appreciate if you can share your experience with me, or
pin point what's the mistakes I made here.
Thanks in advance.
Regards,
David Loh
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