Hi:
I use t_uac_dlg send message.
the destination user in my location table. actually，this user is
offline and not send unregister。
then, I sent lots of message to this user, only previous message can
trigger on_failure_route[].
some log like this:
Apr 14
in my understanding: mi interface will first enter local_route,
in local route , will generate SIP Message, this SIP Message will route
to main REQUEST_ROUTE.
in REQUEST_ROUTE, I have some code part like this:
t_on_failure(sip_message);
if this transaction have any
Hi ALL:
User A sent register to opensips. then User A client core-dump, so
User A will never sent unregister message to opensips.
in this case , how do I Immediately know User A unreachable .
registered function can not know this,
by the way, transport protocol is TCP.
Hello,
Is the leak you are reporting happening in shared memory, or pkg ? Your
logs contain just PKG memory dump ?
OpenSIPS 1.9 is no longer supported ( see [1] ), so I would advise you
to update to the latest OpenSIPS 1.11 ( latest minor release is 1.11.4 )
[1]
Hello,
What you can do is send the call to a destination which is not available
at all, control the amount of time you want to give the client to
register via the fr_timer, and when that timeout is exceeded try to
route the call to the client.
Short snippet of code would be
if
Hello,
From you ul show output, you don't have to do anything special for this
to work - OpenSIPS will automatically relay the call to the Received::
value that's displayed in the ul show output, setting it as $du, while
the actual Request-URI of the message will contain the private Contact
Hello,
If the destination is TCP, then you can detect the failure to relay in
your OpenSIPS script. See the flags that t_relay() takes at
www.opensips.org/html/docs/modules/1.11.x/tm#id294528
if (!t_relay(0x02)) {
# failure to relay , treat error here
}
Make sure to also set low
Thanks Vlad,
$du did magic, it is extracting Received:: value.
On Tue, Apr 14, 2015 at 11:34 AM, Vlad Paiu vladp...@opensips.org wrote:
Hello,
From you ul show output, you don't have to do anything special for this to
work - OpenSIPS will automatically relay the call to the Received::
Hi,
I have following User registred over public IP but that client doesn't
support STUN so contact info showing private IP 192.168.1.6
lookup function default extract Contact:: sip:1001@192.168.1.6:27098
Is there a way i can extract Received:: sip:173.XX.XX.215:27098 so i
can create new URI
Hello, All!
The winner of the third Bug Hunt edition[1] is Bernard
Buitenhuis(@bbuitenhuis)[2] for the fork=no crash. Many thanks for
reporting it!
The fourth edition has now started and lasts until 20th of April, 2015!
Don't miss your chance to win a nice OpenSIPS t-shirt, as well as the
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