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<tt>Hi Ben,</tt><tt><br>
</tt><tt><br>
</tt><tt>Just for the sake of completeness, <tt>if we do d<tt>ecide
to re<tt>tr<tt>y a reques<tt>t within a failure route, we will
not do a "retransmission<tt>", but <tt>actually create
a bran<tt>d </tt>new transaction (with <tt>a</tt>
unique ";branch=" param)<tt> o<tt>n the SIP proxy's
UAC side (the outgoing side)</tt>.<br>
<br>
<tt>Best regards,</tt><br>
</tt></tt></tt></tt></tt></tt></tt></tt><tt></tt></tt><br>
<pre class="moz-signature" cols="72">Liviu Chircu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<div class="moz-cite-prefix">On 17.07.2018 15:54, Ben Newlin wrote:<br>
</div>
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<div class="WordSection1">
<p class="MsoNormal">Gerwin,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Specifically, it is the t_relay() call
within your failure_route. t_relay() is used to send
requests, not responses. The automatic action of failure_route
if no retransmission is attempted is to send the reply back
upstream [1]. So you don’t have to do anything if that is the
functionality you desire.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">[1] <a
href="http://www.opensips.org/Documentation/Script-Routes-2-4#toc3"
moz-do-not-send="true">
http://www.opensips.org/Documentation/Script-Routes-2-4#toc3</a><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="color:black">Ben Newlin </span><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div style="border:none;border-top:solid #B5C4DF
1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span
style="font-size:12.0pt;color:black">From: </span></b><span
style="font-size:12.0pt;color:black">Users
<a class="moz-txt-link-rfc2396E" href="mailto:users-bounces@lists.opensips.org"><users-bounces@lists.opensips.org></a> on behalf of
Liviu Chircu <a class="moz-txt-link-rfc2396E" href="mailto:liviu@opensips.org"><liviu@opensips.org></a><br>
<b>Reply-To: </b>OpenSIPS users mailling list
<a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a><br>
<b>Date: </b>Tuesday, July 17, 2018 at 8:50 AM<br>
<b>To: </b><a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org">"users@lists.opensips.org"</a>
<a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a><br>
<b>Subject: </b>Re: [OpenSIPS-Users] Multiple branches
for a single AOR<o:p></o:p></span></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<p><tt><span style="font-size:10.0pt">Hi Gerwin,</span></tt><o:p></o:p></p>
<p><tt><span style="font-size:10.0pt">Inside your failure route,
you are always attempting a retry of any failed request.
This logic conflicts with your initial statement that "The
intent is to ensure that the response code gets sent
through to A-party".</span></tt><o:p></o:p></p>
<p><tt><span style="font-size:10.0pt">Best regards,</span></tt><o:p></o:p></p>
<pre>Liviu Chircu<o:p></o:p></pre>
<pre>OpenSIPS Developer<o:p></o:p></pre>
<pre><a href="http://www.opensips-solutions.com" moz-do-not-send="true">http://www.opensips-solutions.com</a><o:p></o:p></pre>
<div>
<p class="MsoNormal">On 17.07.2018 14:35, Gerwin van de Steeg
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<p class="MsoNormal">Folks,<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">I'm trying to narrow down a 482
Merged Request problem on calls from one SIP device to
another via OpenSIPS 2.4.1. Yealink T41P SIP device
(A-party), calls via OpenSIPS, to another AOR owned by a
Zoiper5 device (B-party).<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">The intent is to ensure that when the
B-party rejects the call with a 486 Busy Here, that the
response code gets sent through to A-party. However
what I'm seeing is the 486 gets sent to OpenSIPS which
ACK's it, but doesn't go anywhere from there, and then
something causes a second invite to be sent from
OpenSIPS to the B-party which then responds of course
with 482 Merged Request.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal">The call as it is progressing through
the call flow seems to be starting a second branch to
the AOR (only one SIP device registered using UDP per
AOR).<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">What would be causing that second
call so that I can eliminate it and get to the behaviour
I'm expecting. Just using a slightly modified
residential default config template with websocket
support (the problem was noticed using SIP.JS but exists
also in generic SIP device to SIP device calls).<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">Image containing sngrep of call: <a
href="https://imgur.com/RCZXkO6"
moz-do-not-send="true">
https://imgur.com/RCZXkO6</a><o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">Subscribers are in the form of
<username>@<domain><br>
With an alias setup for an extension number.<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">ie.<br>
alfred.anderson@... = 552<o:p></o:p></p>
</div>
<div>
<div>
<p class="MsoNormal"><span style="font-size:12.0pt">
alice.bell@... = 553<o:p></o:p></span></p>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal">excerpt from opensips.cfg<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<p class="MsoNormal"><span style="font-family:"Courier
New""> if ($rU==NULL) {<br>
# request with no Username in RURI<br>
send_reply("484","Address Incomplete");<br>
exit;<br>
}<br>
<br>
$acc_extra(src_ip) = $si; # source IP of the
request<br>
$acc_leg(caller) = $fu;<br>
$acc_leg(callee) = $ru;<br>
<br>
# apply DB based aliases<br>
if (alias_db_lookup("dbaliases")) {<br>
xlog("Alias lookup success
[$fu/$tu/$ru/$ci]");<br>
}<br>
else { <br>
xlog("Alias lookup failure
[$fu/$tu/$ru/$ci]");<br>
}<br>
<br>
# do blind callforward lookup<br>
if (avp_db_load("rU", "$avp(callfwd)")) {<br>
t_reply("181", "Call Is Being
Forwarded");<br>
$ru = $avp(callfwd);<br>
xlog("forwarded call to: $avp(callfwd)");<br>
route(relay);<br>
exit;<br>
}<br>
<br>
# apply transformations from dialplan table<br>
dp_translate("0", "$rU/$rU");<br>
<br>
# check if the call needs to be routed to
freeswitch<br>
route(freeswitch);<br>
<br>
# here we would set the redirect URI if it had
one<br>
route(lookup);<br>
}<br>
<br>
route[lookup] {<br>
</span><span
style="font-size:12.0pt;font-family:"Courier
New";background:white"> script_trace(1, "$rm
from $si, rur=$ru", "me");</span><span
style="font-size:12.0pt;font-family:"Courier
New""><br>
</span><span style="font-family:"Courier New"">
xlog("route:lookup");<br>
# do lookup with method filtering<br>
if (!lookup("location","m")) {<br>
xlog("lookup failure");<br>
t_newtran();<br>
if (!db_does_uri_exist()) {<br>
xlog("$cfg_line: URI doesn't
exist");<br>
send_reply("420", "Bad
Extension");<br>
exit;<br>
}<br>
t_reply("404", "Not Found");<br>
exit;<br>
}<br>
<br>
# when routing via usrloc, log the missed calls
also<br>
do_accounting("db","missed");<br>
<br>
route(relay);<br>
}<br>
<br>
route[freeswitch] {<br>
xlog("route:freeswitch");<br>
if (!is_method("INVITE")) {<br>
return;<br>
}<br>
<br>
# if the called number begins with the right
dialplan redirect it to freeswitch<br>
# here we take everythign prefixed with a *,
strip it, and send it to freeswitch<br>
if ($rU=~"^\*") {<br>
strip(1);<br>
$du = "sip:<a
href="http://10.23.4.192:50600" moz-do-not-send="true">10.23.4.192:50600</a>";<br>
route(relay);<br>
}<br>
}<br>
<br>
<br>
route[relay] {<br>
xlog("route:relay: Relaying: method=$rm");<br>
# for INVITEs enable some additional helper
routes<br>
if (is_method("INVITE")) {<br>
t_on_branch("per_branch_ops");<br>
t_on_reply("handle_nat");<br>
t_on_failure("missed_call");<br>
}<br>
else if (is_method("BYE|CANCEL")) {<br>
# cancel the rtpengine transcoding<br>
rtpengine_delete();<br>
}<br>
<br>
if (!t_relay()) {<br>
send_reply("500","Internal Error");<br>
}<br>
exit;<br>
}<br>
<br>
<br>
branch_route[per_branch_ops] {<br>
script_trace(1, "$rm from $si, rur=$ru", "me");<br>
xlog("[$ci/$T_branch_idx]
branch_route:per_branch_ops: new branch at $ru\n");<br>
<br>
# WebSocket specific handling with NORMAL SDP
negotiation<br>
# assumes SDP offer in the INVITE from the UAC,
and SDP<br>
# answer is in 200 OK from the UAS<br>
if (!is_method("INVITE") ||
!has_body("application/sdp"))<br>
return;<br>
<br>
if (isflagset(SRC_WS) &&
isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "ICE=force-relay
DTLS=passive";<br>
else if (isflagset(SRC_WS) &&
!isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "RTP/AVP
replace-session-connection replace-origin ICE=remove";<br>
else if (!isflagset(SRC_WS) &&
isbflagset(DST_WS))<br>
$var(rtpengine_flags) =
"UDP/TLS/RTP/SAVPF ICE=force";<br>
else if (!isflagset(SRC_WS) &&
!isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "RTP/AVP
replace-session-connection replace-origin ICE=remove";<br>
<br>
# only enable transcoding if websocket call for
now<br>
if (isflagset(SRC_WS) || isbflagset(DST_WS)) {<br>
rtpengine_offer("$var(rtpengine_flags)");<br>
}<br>
}<br>
<br>
onreply_route[handle_nat] {<br>
script_trace(1, "$rm from $si, rur=$ru", "me");<br>
xlog("[$ci/$T_branch_idx]
onreply_route:handle_nat: $ru\n");<br>
<br>
# WebSocket specific handling with NORMAL SDP
negotiation<br>
# assumes SDP offer in the INVITE from the UAC,
and SDP<br>
# answer is in 200 OK from the UAS<br>
if (!has_body("application/sdp"))<br>
return;<br>
<br>
if (isflagset(SRC_WS) &&
isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "ICE=force-relay
DTLS=passive";<br>
else if (isflagset(SRC_WS) &&
!isbflagset(DST_WS))<br>
$var(rtpengine_flags) =
"UDP/TLS/RTP/SAVPF ICE=force";<br>
else if (!isflagset(SRC_WS) &&
isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "RTP/AVP
replace-session-connection replace-origin ICE=remove";<br>
else if (!isflagset(SRC_WS) &&
!isbflagset(DST_WS))<br>
$var(rtpengine_flags) = "RTP/AVP
replace-session-connection replace-origin ICE=remove";<br>
<br>
# only enable transcoding if websocket call for
now<br>
if (isflagset(SRC_WS) || isbflagset(DST_WS)) {<br>
rtpengine_answer("$var(rtpengine_flags)");<br>
}<br>
}<br>
<br>
failure_route[missed_call] {<br>
script_trace(1, "$rm from $si, rur=$ru", "me");<br>
xlog("[$ci/$T_branch_idx]
failure_route:missed_call: incoming failure response to
$rm <- $T_reply_code/$T_ruri");<br>
if (t_was_cancelled()) {<br>
xlog("[$ci/$T_branch_idx] was
cancelled");<br>
exit;<br>
}<br>
do_accounting("db", "missed");<br>
<br>
if (!t_relay()) {<br>
send_reply("500","Internal Error");<br>
}<br>
else {<br>
xlog("[$ci/$T_branch_idx] Relay success
$rm/$T_reply_code");<br>
}<br>
}</span> <o:p></o:p></p>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal"><o:p> </o:p></p>
</div>
<p class="MsoNormal"><br>
Cheers,<br>
Gerwin <o:p></o:p></p>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span
style="font-family:"Courier New""><br>
<br>
</span><o:p></o:p></p>
</div>
</div>
</div>
<p class="MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Users mailing list<o:p></o:p></pre>
<pre><a href="mailto:Users@lists.opensips.org" moz-do-not-send="true">Users@lists.opensips.org</a><o:p></o:p></pre>
<pre><a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" moz-do-not-send="true">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><o:p></o:p></pre>
</blockquote>
<p class="MsoNormal"><br>
<br>
<o:p></o:p></p>
</div>
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<pre wrap="">_______________________________________________
Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
<a class="moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a>
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