Hello,
Looking in your SIP trace, I see that in the 200OK Contact, you have
Contact: <sip:72.XX.XX.140;did=7de.9accc6f5>. , and when OpenSIPS is
routing the ACK out, it is routing it to
U 182.XX.XX.164:5060 -> 72.XX.XX.140:5060
ACK sip:72.XX.XX.140;did=7de.9accc6f5 SIP/2.0.
so not sure where exactly is the loop.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07.04.2015 19:32, Satish Patel wrote:
It is URGENT!!
can some one help? This is very strange issue and i am stuck here :(
loose_route() sending ACK/BYE itself instead of next hope :(
I have removed all entries from "domain" table but no luck :(
On Thu, Mar 26, 2015 at 12:09 AM, Satish Patel <[email protected]
<mailto:[email protected]>> wrote:
Hi,
senario:
[UA]-------------[Opensips]---------[Freeswitch]
UA sending correct ACK to freeswitch but Opensips loose_route()
sending it to itself and it break dialog, If use
fix_dialog_route() then it works, I don't have any IP address
added in domain table also.
How do i check whether Freeswitch using loose_route for strict route?
I have following script:
if (has_totag()) {
if (loose_route()) {
if (is_method("BYE")) {
#setflag(ACC_DO); # do accounting ...
#setflag(ACC_FAILED); # ... 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();
}
if (check_route_param("nat=yes"))
setflag(NAT);
# route it out to whatever destination was
set by loose_route()
# in $du (destination URI).
route(relay);
} 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. 404 from
upstream server
xlog("non loose-route section\n");
#t_relay();
exit;
} else {
# ACK without matching
transaction ->
# ignore and discard
xlog("ACK without matching transaction\n");
exit;
}
}
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