Hi,
just a quick note to tell you that I actually thought about this way of
using fix_route_dialog() because of this ticket:
http://sourceforge.net/tracker/?func=detail&aid=3090388&group_id=232389&atid=1086413
You may want to correct it?
Guido
On 25/01/11 11:14, Vlad Paiu wrote:
Hi,
Are you sure you are calling record_route() properly in your script ?
If you do this, you shouldn't receive any SIP msgs without any route
headers, as there would be at least 1 route header, the one OpenSIPS
has added in the first place.
The dialog module depends on the RR module. If a SIP msg with no route
headers is received, the loose_route() part rejects it as it isn't
valid from it's point of view, and the message isn't attempted to be
matched to any certain dialog.
Could you please post a full SIP trace for such a call ?
Regards,
Vlad
On 01/24/2011 06:35 PM, Guido Negro wrote:
Hi,
thanks for your reply.
I am using the script reported in the Opensips book ("Building
telephony systems with Opensips") at page 81, that is :
if (has_totag() ) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route() ) {
if (is_method("BYE" ) ) {
setflag(1) ; # do accounting . . .
setflag(3 ) ; # . . . even if the transaction fails
} else if ( is_method( "INVITE" ) ) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route() ;
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI) .
route( 1) ;
} else {
if ( is_method(" ACK") ) {
if ( t_check_trans( ) ) {
# non loose- route, but stateful ACK; must be an ACK after
# a 487 or e. g. 4 04 from upstream server
t_relay() ;
exit;
} else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
}
sl_send_reply("4 04" , " Not here") ;
}
exit;
}
Now I've got 2 problems. When a BYE with correct to/from tag &
correct call-id, but without any route headers arrives (a message I'd
like to 'fix'):
1) I get the variable $DLG_status to be <null> after loose_route() (I
checked with specific experiments) and this also causes an error in
the log: opensips[3216]: ERROR:dialog:w_validate_dialog: null dialog
2) loose_route() fails, so I always get a 404 response on the callee
side that tried to disconnect, and the caller remains connected.
I am probably using fix_route_dialog not in the correct way, but what
I wanted to do is something like this (very-pseudo code):
When loose_route fails (or more in general for every in-dialog
subsequent request):
Try to fix the request according to saved dialog infos:
- Fix succesful: forward the request.
- Couldnt fix it: reject the request (send 404 back).
Regards,
Guido.
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