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<tt>Marco,<br>
<br>
<tt>In this is case it <tt>exclusively</tt><tt><tt> depends on
the <tt>rtpe<tt>ngine if <tt>it <tt>update<tt>s or
creates a new session when you call the "offer"
for the second time. I<tt>f th<tt>e ex<tt>isting <tt>session
(created for the first branch) will be
updated on the secon<tt>d "offer", the po<tt>rt
in the replies <tt>(<tt>towards
caller) will not change, right ?<br>
<br>
<tt>B<tt>ut, based on your tests,
I assume i<tt>t does ch<tt>ange
the port send to caller
after the second offer :(
.<br>
<br>
<tt>Again, this depends on
the media relay
implementation, it <tt>is
not something to be
controlled by
OpenSIPS.<br>
<br>
<tt>As a<tt> simple
alternative, move
<tt>your media
server on the
public network,
so <tt>you will
not need a s<tt>econ<tt>d
offer. For the
second branch,
you <tt>will
simply do a<tt>
second answer<tt>.</tt></tt></tt></tt></tt></tt></tt></tt></tt><br>
</tt></tt></tt></tt></tt></tt><br>
<tt>Regards,</tt><br>
</tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></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>
</pre>
<div class="moz-cite-prefix">On 11/22/2017 06:37 PM, Marco Hierl
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:08c1e9f75c2f449d9bc1273182f9b526@next-id.de">
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<div class="WordSection1">
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">Hi
Bogdan,<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">yes,
that’s correct!<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"><o:p> </o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">Best
regards<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> Marco<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"><o:p> </o:p></span></p>
<div>
<div style="border:none;border-top:solid #B5C4DF
1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span
style="font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:DE">Von:</span></b><span
style="font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:DE">
Bogdan-Andrei Iancu [<a class="moz-txt-link-freetext" href="mailto:bogdan@opensips.org">mailto:bogdan@opensips.org</a>]
<br>
<b>Gesendet:</b> Mittwoch, 22. November 2017 16:15<br>
<b>An:</b> Marco Hierl; OpenSIPS users mailling list<br>
<b>Betreff:</b> Re: AW: [OpenSIPS-Users] rtp port
changed in REPLY's SDP after announcement<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><tt>Hi Marco,</tt><span
style="font-family:"Courier
New","serif""><br>
<br>
<tt>If I understand correctly, you *NEED* to do get
different SDP's in the 2 INVITE branches as the first
branch going to the media server needs to have a private
IP, while the second branch going to the calle needs to
have a public IP, right ?</tt><br>
<br>
<tt>Regards,</tt><br>
<br>
</span><o:p></o:p></p>
<pre>Bogdan-Andrei Iancu<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>
<div>
<p class="MsoNormal">On 11/21/2017 12:49 PM, Marco Hierl
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D">Hi Bogdan,</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">no, he
did not say anything yet, but I expect that it is
paragraph 3.2:</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB">“A UAS cannot send a
new offer in the reliable</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB">provisional response,
so the UPDATE method is the only method for a</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB">UAS to update an early
session.”</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">(I’m
not quite sure if this really fits to our situation,
because the UAS itself changed.)</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">According
to your suggestion:
</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">I used
“</span><span style="font-family:"Courier New ,
serif","serif"" lang="EN-GB">rtpengine_offer(external
</span><b><span style="font-family:"Courier New , serif
;color:red","serif"" lang="EN-GB">internal</span></b><span
style="font-family:"Courier New ,
serif","serif"" lang="EN-GB">…”
</span><span style="font-size:11.0pt;color:#1F497D"
lang="EN-GB">to adjust the SDP the first time. So I get an
internal IP in the INVITE to the announcement. And this
will be seen in the 2<sup>nd</sup> INVITE to the callee if
I do NOT make “</span><span
style="font-family:"Courier New ,
serif","serif"" lang="EN-GB">rtpengine_offer(external
</span><b><span style="font-family:"Courier New , serif
;color:red","serif"" lang="EN-GB">external</span></b><span
style="font-family:"Courier New ,
serif","serif"" lang="EN-GB">…</span><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">” to
the INVITE to the callee. But the callee has public IP and
I need to use the external interface.
</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">Thanks
for your help!</span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB">
Marco</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;color:#1F497D" lang="EN-GB"> </span><o:p></o:p></p>
<div>
<div style="border:none;border-top:solid #B5C4DF
1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span
style="font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:DE">Von:</span></b><span
style="font-family:"Tahoma","sans-serif";color:windowtext;mso-fareast-language:DE">
Bogdan-Andrei Iancu [<a
href="mailto:bogdan@opensips.org"
moz-do-not-send="true">mailto:bogdan@opensips.org</a>]
<br>
<b>Gesendet:</b> Dienstag, 21. November 2017 11:10<br>
<b>An:</b> OpenSIPS users mailling list; Marco Hierl<br>
<b>Betreff:</b> Re: [OpenSIPS-Users] rtp port changed
in REPLY's SDP after announcement</span><o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><tt>Hello Marco,</tt><span
style="font-family:"Courier New ,
serif","serif""><br>
<br>
</span><tt>Did the interconnection partner mentioned the
chapter/paragraph in RFC6337 where the restriction on the
port is mentioned (I was not able to find it on a quick
scan) ? it is the first time I hear about anything like
this.</tt><span style="font-family:"Courier New ,
serif","serif""><br>
<br>
</span><tt>Now, if you use rtpproxy (I do not know about
rtpengine) on your side and you do rtpproxy_offer only
once (when getting the call from the carrier) , you can do
rtpproxy_answer multiple time (during the serial forking).
In this case, the RTP IP:port towards the carrier will
never change, but the RTP callee leg will be changed.</tt><span
style="font-family:"Courier New ,
serif","serif""><br>
<br>
</span><tt>Best regards,</tt><span
style="font-family:"Courier New ,
serif","serif""><br>
<br>
<br>
</span><o:p></o:p></p>
<pre>Bogdan-Andrei Iancu<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>
<div>
<p class="MsoNormal">On 11/21/2017 11:07 AM, Marco Hierl
wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">Dear all,</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">I want to play an early announcement before
connecting a call. The announcement can be heard by the
caller, but the callee can't hear the caller after the
call is connected!</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">To be more precise:</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">The openSIPS proxy receives a call from an
external interconnection partner (public IP on external
interface) and sends it on to an announcement server
(private IP on internal interface).
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">After REPLY 183 the announcement starts and
the caller can hear the announcement. It finishes with
REPLY603 that comes to the openSIPS failure route. Some
stuff is done (e.g. changing R-URI),
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">a new t_relay will be done and an INVITE is
send to the callee (public IP on external interface).
The callee sends 200ok, it's forwarded to the caller,
that sends ACK like usual.
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">But the caller does not send RTP to the
IP/PORT indicated in the 200OK.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">The interconnection partner said that the
problem is, that the RTP port in REPLY 183 is different
to that in REPLY200 (not allowed, see RFC6337). (just
to mention: the to-tag is different too)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">I'm using openSIPS 2.3 and in order to
overcome the issue with internal and external addresses
and also to keep the rtp streams I'm using rtpengine 5.4
(on a different machine)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">Is the change of the RTP Port allowed in
this way? If not, what I can do better? Or is it a fault
from rtp-engine, should it keep the first port?</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">Any help is very appreciated!</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">I tried to make a picture... maybe it helps
a bit to understand what I was not able to explain
above...</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">No
From To Message</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">1
external ownProxy Request: INVITE
<a href="sip:+49xxx@us.de;user=phone"
moz-do-not-send="true">sip:+49xxx@us.de;user=phone</a></span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">2
ownProxy external Status: 100 Giving a try</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> rtpengine_offer(external
internal replace-session-connection ICE=remove
trust-address replace-origin)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 3
to
announcement server Request: INVITE
<a href="sip:ANNC@mediaserver:5060;user=phone"
moz-do-not-send="true">sip:ANNC@mediaserver:5060;user=phone</a>
|
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 4
from announcement server Status: 100 Trying |
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 5
from announcement server
Status: 183 Session Progress |
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> o=root
1135152900 1135152900 IN IP4 mediaserver</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
m=audio 13788 RTP/AVP 8 101</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
rtpengine_answer(internal external
replace-session-connection ICE=remove trust-address
replace-origin)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 6
ownProxy external Status: 183 Session Progress
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> o=root
1135152900 1135152900 IN IP4 ownProxy</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
m=audio 23024 RTP/AVP 8 101</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> <
========== ANNOUNCEMENT =========
>
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 7
from announcement server Status: 603 Declined
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
openSIPS failure route
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> and
rtpengine_offer(external external
replace-session-connection ICE=remove trust-address
replace-origin)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
and new t_relay</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> 8
to callee Request: INVITE
<a href="sip:bob@externalcarrier:5060;user=phone"
moz-do-not-send="true">sip:bob@externalcarrier:5060;user=phone</a></span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">9
from callee Status: 100 Trying</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">10
from callee Status: 200 OK</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
o=-
239036129 823630844 IN IP4 10.0.203.7</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
m=audio 1044 RTP/AVP 8 101</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
rtpengine_answer(external external
replace-session-connection ICE=remove trust-address
replace-origin)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">11
ownProxy external Status: 200 OK</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
o=- 239036129 823630844 IN IP4 ownProxy</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">
m=audio 23056 RTP/AVP 8 101
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> <
========== RTP from B to A is ok, A send NO RTP to
B =========
>
</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">12
external ownProxy Request: ACK
<a href="sip:ownProxy;did=5db.a2833e17"
moz-do-not-send="true">sip:ownProxy;did=5db.a2833e17</a></span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">13
external ownProxy Request: BYE
<a href="sip:ownProxy;did=5db.a2833e17"
moz-do-not-send="true">sip:ownProxy;did=5db.a2833e17</a></span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-family:"Courier
New , serif","serif"" lang="EN-GB">14
ownProxy external Status: 200 OK</span><o:p></o:p></p>
<p class="MsoNormal"><span lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">version: opensips 2.3.0 (x86_64/linux)</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">flags: STATS: On, DISABLE_NAGLE, USE_MCAST,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">ADAPTIVE_WAIT_LOOPS=1024,
MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE
1024, BUF_SIZE 65535</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">poll method support: poll, epoll_lt,
epoll_et, sigio_rt, select.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">git revision: fe0ebcb45</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">main.c compiled on 09:02:02 Sep 14 2017
with gcc 6.3.0</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB">rtpengine version 5.4.0.0+0~mr5.4.0.0
git-master-a9d55c6</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"
lang="EN-GB"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt">Thanks
and best regards!!!</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> Marco</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:11.0pt"> </span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size:12.0pt"><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="font-size:12.0pt"> </span><o:p></o:p></p>
</blockquote>
<p class="MsoNormal"><span
style="font-size:12.0pt;font-family:"Times New
Roman","serif";mso-fareast-language:DE"><o:p> </o:p></span></p>
</div>
</blockquote>
<br>
</body>
</html>