Hi Klaus,
The $ru trick works perfectly (without resetdsturi()) !
Now the disconnection (hang up) is handled properly.
Thank a million.
Regards,
David Loh
Klaus Darilion wrote:
Klaus Darilion schrieb:
David Loh schrieb:
Hi Klaus,
So in order to make it work, the RURI of Asterisk uses should
contain "transport=TLS" right.
yes
if the "transport=TLS" can be appended to the SIP message, the
disconnection shall be handle properly ?
yes
Currently I'm struggling w/ subst/subst_uri ... it's seems the Regex
textops module used was slightly different from Unix,
I do "subst('/^BYE(.*)SIP\/2\.0/BYE\1;transport=TLS SIP\/2\.0/ ');"
but it doesn't work ...
I'm not sure if subst able to alter the header but if it doesn't, is
there any command that I can use to alter the BYE header ?
There is no need to use subst - just rewrite the request URI. E.g. in
openser 1.2 the following should work:
if (loose_route()) {
...
if (src_ip == ip.address.of.asterisk) {
$ru = $ru + ";transport=tls";
I do not know for sure, but maybe it is necessary to reset the duri
(may be set during loose_route()):
resetdsturi();
}
...
t_relay();
exit;
}
regards
klaus
Thanks,
David Loh
Klaus Darilion wrote:
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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