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<tt>Hi Brian,<br>
<br>
<tt>Keep in mind that you cannot make opensips act in the same
time as proxy (as required by the load balancer) a<tt>nd as a
end-point (as required by the B2BUA). Id<tt>eally is to r<tt>un
the <tt>two <tt>services (<tt>LB and B2B) on t<tt>wo
opensips <tt>instances<tt> in a chain.</tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><br>
<br>
<tt>Best regards,</tt><br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
OpenSIPS Summit 2018
<a class="moz-txt-link-freetext" href="http://www.opensips.org/events/Summit-2018Amsterdam">http://www.opensips.org/events/Summit-2018Amsterdam</a>
</pre>
<div class="moz-cite-prefix">On 02/02/2018 07:03 PM, Brian
Southworth wrote:<br>
</div>
<blockquote type="cite"
cite="mid:01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com">
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<p class="MsoNormal"><span style="color:windowtext">Sorry my
apologies.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">So from the
beginning opensips acts as an authorization proxy which
passes the call on to an asterisk box based on load (using
load balancer).<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">I am trying
to get the opensips proxy to handle call transfers and I
thought the b2bua would be the best way. Initially the refer
was sent to the asterisk box.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">On inbound
calls <o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">The call
comes in from the carrier goes to asterisk, asterisk then
passes the sip invite to the proxy which then rings the sip
phone.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">What I wish
to achieve is a way to transfer an inbound call to an
internal extension or external number.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">Example: <o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">Caller A
receives call </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> caller A places call on hold and
dials caller B </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> caller B picks up </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> caller A presses cisco xfer and
call is passed to caller B<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">I was hoping
to achieve this using the proxy or asterisk box if possible.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext">I hope this
helps.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:12.0pt;mso-fareast-language:EN-GB">Regards,<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB">Brian
Southworth<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:windowtext"><o:p> </o:p></span></p>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US">From:</span></b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US"> Bogdan-Andrei Iancu
[<a class="moz-txt-link-freetext" href="mailto:bogdan@opensips.org">mailto:bogdan@opensips.org</a>] <br>
<b>Sent:</b> 02 February 2018 16:50<br>
<b>To:</b> Brian Southworth
<a class="moz-txt-link-rfc2396E" href="mailto:brian.southworth@clocom.uk"><brian.southworth@clocom.uk></a>; OpenSIPS users
mailling list <a class="moz-txt-link-rfc2396E" href="mailto:users@lists.opensips.org"><users@lists.opensips.org></a><br>
<b>Subject:</b> Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><tt><span style="font-size:10.0pt">I'm a
bit confused. The original report was on a record_route()
/ loose_route() matter. But you say you have opensips as
B2B, so the RR mechanism must not be used in such a case -
you act either as a end-point, either as a proxy - you
cannot be both for the same call.</span></tt><span
style="font-size:10.0pt;font-family:"Courier
New",serif"><br>
<br>
<tt>Now you have this b2b error, during a call transfer
scenario. and you mentioned LB also :)...so I'm a bit
confused - could please try to put all these pieces
together, so I can understand what you are doing ?</tt><br>
<br>
<tt>Regards,</tt><br>
<br>
</span><span style="mso-fareast-language:EN-GB"><o:p></o:p></span></p>
<pre>Bogdan-Andrei Iancu<o:p></o:p></pre>
<pre><o:p> </o:p></pre>
<pre>OpenSIPS Founder and 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>
<pre>OpenSIPS Summit 2018<o:p></o:p></pre>
<pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam" moz-do-not-send="true">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre>
<div>
<p class="MsoNormal">On 02/02/2018 04:27 PM, Brian Southworth
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="color:windowtext">Maybe I am
doing this wrong but I wanted the B2BUA module to handle
the refer and bridge the calls. </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">I have the
B2bUA working now. However my issue is that its not able
to send the replies.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">incoming
reply</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">b2b_reply
(B2B.222.7591351.1517580641)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Feb 2
14:10:47 [22664] ERROR:tm:_reply_light: failed to generate
408 reply when a final 200 was sent out</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Feb 2
14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed
to send reply with tm</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Feb 2
14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply:
Sending reply failed - 408, [B2B.452.342.1517580641]</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Do you
need anything else to help me debug this ? I am not sure
why its failing to pass the reply with tm, I have enabled
the param:</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:10.0pt">modparam("tm",
"pass_provisional_replies", 1)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">I should
also note that I am using the load balancer module also.
This normally deals with all call distribution. In and
out.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<div>
<p class="MsoNormal"><span
style="font-size:12.0pt;mso-fareast-language:EN-GB">Regards,</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB">Brian
Southworth</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
</div>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US">From:</span></b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US"> Bogdan-Andrei Iancu [<a
href="mailto:bogdan@opensips.org"
moz-do-not-send="true">mailto:bogdan@opensips.org</a>]
<br>
<b>Sent:</b> 02 February 2018 14:20<br>
<b>To:</b> Brian Southworth <a
href="mailto:brian.southworth@clocom.uk"
moz-do-not-send="true"><brian.southworth@clocom.uk></a>;
OpenSIPS users mailling list <a
href="mailto:users@lists.opensips.org"
moz-do-not-send="true"><users@lists.opensips.org></a><br>
<b>Subject:</b> Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]</span><o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><tt><span style="font-size:10.0pt">Hi
Brian,</span></tt><span
style="font-size:10.0pt;font-family:"Courier
New",serif"><br>
<br>
<tt>Maybe that warning points to a routing error that
prevents the REFER to be route to carrier - make a sip
capture to be sure the REFER from A is properly routed
and accepted by the carrier.</tt><br>
<br>
<tt>Regards,</tt><br>
<br>
<br>
</span><o:p></o:p></p>
<pre>Bogdan-Andrei Iancu<o:p></o:p></pre>
<pre> <o:p></o:p></pre>
<pre>OpenSIPS Founder and 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>
<pre>OpenSIPS Summit 2018<o:p></o:p></pre>
<pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam" moz-do-not-send="true">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre>
<div>
<p class="MsoNormal">On 02/02/2018 01:38 PM, Brian
Southworth wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="color:windowtext">Hi
Bogdan,</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Thank
you very much, so this doesn’t have any impact on why
the call being transferred are dropped ?</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">I am
trying to transfer a call using the refer method as that
is what the cisco phones use.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">The
network is setup like so opensips proxy </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> asterisk gateway </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> carrier</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Scenario:</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Inbound
call comes into the phone like so: carrier </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> ast </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> proxy </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> phone A</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Phone A
needs to transfer call to phone B: Phone A dials phone B
</span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> phone B picks up </span><span
style="font-family:Wingdings;color:windowtext">à</span><span
style="color:windowtext"> phone A presses xfer button
and call is dropped.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext">Any help
would be appreciated.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<div>
<p class="MsoNormal"><span
style="font-size:12.0pt;mso-fareast-language:EN-GB">Regards,</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB">Brian
Southworth</span><o:p></o:p></p>
</div>
<p class="MsoNormal"><span style="color:windowtext"> </span><o:p></o:p></p>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US">From:</span></b><span
style="color:windowtext;mso-fareast-language:EN-GB"
lang="EN-US"> Bogdan-Andrei Iancu [<a
href="mailto:bogdan@opensips.org"
moz-do-not-send="true">mailto:bogdan@opensips.org</a>]
<br>
<b>Sent:</b> 02 February 2018 11:29<br>
<b>To:</b> OpenSIPS users mailling list <a
href="mailto:users@lists.opensips.org"
moz-do-not-send="true"><users@lists.opensips.org></a>;
Brian Southworth <a
href="mailto:brian.southworth@clocom.uk"
moz-do-not-send="true"><brian.southworth@clocom.uk></a><br>
<b>Subject:</b> Re: [OpenSIPS-Users] [15066]
WARNING:rr:after_strict: no socket found to match RR
[1][XXX.XXX.XXX.XXX:5060]</span><o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><tt><span style="font-size:10.0pt">Hi
Brian,</span></tt><span
style="font-size:10.0pt;font-family:"Courier
New",serif"><br>
<br>
<tt>That warning means OpenSIPS found a Route header
(while doing loose_route) that is suppose to be of its
own, but the network information from the header does
not match any of the OpenSIPS SIP listeners.</tt><br>
<br>
<tt>Best regards,</tt><br>
<br>
<br>
<br>
</span><o:p></o:p></p>
<pre>Bogdan-Andrei Iancu<o:p></o:p></pre>
<pre> <o:p></o:p></pre>
<pre>OpenSIPS Founder and 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>
<pre>OpenSIPS Summit 2018<o:p></o:p></pre>
<pre> <a href="http://www.opensips.org/events/Summit-2018Amsterdam" moz-do-not-send="true">http://www.opensips.org/events/Summit-2018Amsterdam</a><o:p></o:p></pre>
<div>
<p class="MsoNormal">On 02/02/2018 11:14 AM, Brian
Southworth wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">I get this when trying to transfer
calls using the B2BUA:<o:p></o:p></p>
<p class="MsoNormal">[15066] WARNING:rr:after_strict: no
socket found to match RR [1][xxx.xxx.xxx.xxx:5060]<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">When I try looking on the mailing
list there are no other similar posts, could you please
shed some light on what maybe causing this please.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:12.0pt;mso-fareast-language:EN-GB">Regards,</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:16.0pt;mso-fareast-language:EN-GB">Brian
Southworth</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="mso-fareast-language:EN-GB"><br>
<br>
<br>
<br>
<br>
</span><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"><span
style="mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
</blockquote>
<p class="MsoNormal"><span style="mso-fareast-language:EN-GB"> </span><o:p></o:p></p>
</blockquote>
<p class="MsoNormal"><span style="mso-fareast-language:EN-GB"><o:p> </o:p></span></p>
</div>
</blockquote>
<br>
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