Yes, sometimes there are more than one INVITE (custom platform specific
behavior), in case of 1 INVITE per-dialogue it works nice. Can you suggest
something to cover such cases?
2015-12-15 14:12 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
Maybe there is a parallel forking and one branch gets
timed out (in this
case 408 is selected against 486). How many INVITE requests do you see
being sent out? Or you can eventually make the sip trace available for
viewing on this mailing list or some web site/pastebin out there.
Cheers,
Daniel
On 15/12/15 12:54, Alexandru Covalschi wrote:
I use sngrep to track view the flow and I'm pretty sure it's accurate
enough to tell me that.
Here's relay route:
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route"))
t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route"))
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route"))
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
and here's reply routes
# Manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# Manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
if (t_check_status("486")) {
exit;
}
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
}
However when endpoint replies with 486 BUSY I can't see that on FS,
Kamailio just sends 408 REQ TERM after some amount of time
2015-12-15 13:34 GMT+02:00 Alex Balashov <abalashov(a)evaristesys.com>om>:
From what you describe, the reply should be going
back to the sender. Are
you absolutely sure that it's not? If so, are there any other actions you
could be taking somewhere to drop it, such as in an onreply_route?
ACKs to negative final replies are hop-by-hop, so the ACK you're seeing
directly from the proxy to the UAS is normal.
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web:
http://www.evaristesys.com/, <http://www.csrpswitch.com/>
http://www.csrpswitch.com/
Sent from my BlackBerry.
*From: *Alexandru Covalschi
*Sent: *Tuesday, December 15, 2015 05:03
*To: *Kamailio (SER) - Users Mailing List
*Reply To: *Kamailio (SER) - Users Mailing List
*Subject: *[SR-Users] Relaying failure codes back to initial server
Hello everyone!
I need to relay 486/408/... other failure codes back to initial INVITE
server. Here
<http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html>
http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html
is recommended just to exit failure_route, but that didn't work for me. I
need that to let Freeswitch know which cause has ended the call. Now
Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly
tell me how to achieve that?
Thanks in advance
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web:
http://abs-telecom.com/
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web:
http://abs-telecom.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
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--
Daniel-Constantin
Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio -
http://www.asipto.comhttp://miconda.eu
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--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: