<div dir="ltr">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?<br></div><div class="gmail_extra"><br><div class="gmail_quote">2015-12-15 14:12 GMT+02:00 Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
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.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 15/12/15 12:54, Alexandru Covalschi
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>
<div>I use sngrep to track view the flow and I'm pretty sure
it's accurate enough to tell me that. <br>
</div>
Here's relay route:<br>
route[RELAY] {<br>
<br>
# enable additional event routes for forwarded
requests<br>
# - serial forking, RTP relaying handling, a.s.o.<br>
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {<br>
if(!t_is_set("branch_route"))
t_on_branch("MANAGE_BRANCH");<br>
}<br>
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {<br>
if(!t_is_set("onreply_route"))
t_on_reply("MANAGE_REPLY");<br>
}<br>
if (is_method("INVITE")) {<br>
if(!t_is_set("failure_route"))
t_on_failure("MANAGE_FAILURE");<br>
}<br>
if (!t_relay()) {<br>
sl_reply_error();<br>
}<br>
exit;<br>
}<br>
<br>
</div>
and here's reply routes<br>
<br>
# Manage outgoing branches<br>
branch_route[MANAGE_BRANCH] {<br>
xdbg("new branch [$T_branch_idx] to $ru\n");<br>
route(NATMANAGE);<br>
}<br>
<br>
# Manage incoming replies<br>
onreply_route[MANAGE_REPLY] {<br>
xdbg("incoming reply\n");<br>
if(status=~"[12][0-9][0-9]")<br>
route(NATMANAGE);<br>
}<br>
<br>
# Manage failure routing cases<br>
failure_route[MANAGE_FAILURE] {<br>
<br>
if (t_check_status("486")) {<br>
exit;<br>
}<br>
route(NATMANAGE);<br>
<br>
if (t_is_canceled()) {<br>
exit;<br>
}<br>
<br>
}<br>
<br>
<br>
</div>
<div>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<br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-12-15 13:34 GMT+02:00 Alex
Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="background-color:rgb(255,255,255);line-height:initial">
<div>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?</div>
<div><br>
</div>
<div>ACKs
to negative final replies are hop-by-hop, so the ACK
you're seeing directly from the proxy to the UAS is
normal. <span></span></div>
<div><br>
</div>
<div>--<br>
Alex Balashov | Principal | Evariste Systems LLC<br>
303 Perimeter Center North, Suite 300<br>
Atlanta, GA 30346<br>
United States<br>
<br>
Tel: <a href="tel:%2B1-800-250-5920" value="+18002505920" target="_blank">+1-800-250-5920</a> (toll-free) / <a href="tel:%2B1-678-954-0671" value="+16789540671" target="_blank">+1-678-954-0671</a> (direct)<br>
Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" target="_blank"></a><a href="http://www.csrpswitch.com/" target="_blank">http://www.csrpswitch.com/</a><br>
<br>
Sent from my BlackBerry.</div>
<table style="background-color:white;border-spacing:0px" width="100%">
<tbody>
<tr>
<td colspan="2" style="font-size:initial;text-align:initial;background-color:rgb(255,255,255)">
<div>
<div><b>From: </b>Alexandru Covalschi</div>
<div><b>Sent: </b>Tuesday, December 15, 2015
05:03</div>
<div><b>To: </b>Kamailio (SER) - Users Mailing
List</div>
<div><b>Reply To: </b>Kamailio (SER) - Users
Mailing List</div>
<div><b>Subject: </b>[SR-Users] Relaying
failure codes back to initial server</div>
</div>
</td>
</tr>
</tbody>
</table>
<div>
<div><br>
<div>
<div dir="ltr">Hello everyone!<br clear="all">
<div>I need to relay 486/408/... other failure
codes back to initial INVITE server. Here <a href="http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html" target="_blank"></a><a href="http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html" target="_blank">http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html</a>
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?<br>
</div>
<div>Thanks in advance<br>
</div>
<div>-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
</div>
<br>
</div>
</div>
</div>
</div>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
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<br>
</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
<div>
<div dir="ltr">Alexandru Covalschi<br>
ABRISS-Solutions
<div>VoIP engineer and system administrator<br>
phone: <a href="tel:%2B37367398493" value="+37367398493" target="_blank">+37367398493</a><br>
web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div>
</div>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
<pre>_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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</pre>
</blockquote>
<br>
</div></div><span class="HOEnZb"><font color="#888888"><pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://miconda.eu" target="_blank">http://miconda.eu</a></pre>
</font></span></div>
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature"><div dir="ltr">Alexandru Covalschi<br>ABRISS-Solutions<div>VoIP engineer and system administrator<br>phone: +37367398493<br>web: <a href="http://abs-telecom.com/" target="_blank">http://abs-telecom.com/</a></div></div></div>
</div>