<div dir="ltr"><img width="0" height="0" class="mailtrack-img" style="float:right;" alt="" src="https://mailtrack.io/trace/mail/3b2f84b07dfe86fc82224abf7b2cd58ee29ab86a.png?u=1422671">As you may have noticed in my last reply, I reached that far as well but got stuck later on on what appears to be the rtp engine configuration.<div><br></div><div>Not strictly an Opensips issue but you might be able to help me.<br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<tt>Hi Alex,<br>
<br>
Thank you for the offlist provided data. Shortly, the ACK received
by OpenSIPS from OmniPCX is broken as it is missing all the Route
headers. According to the pcap, it looks like:<span class=""><br>
<br>
ACK <a class="m_-865090424726876519moz-txt-link-freetext">sip:udoioiia@10.0.1.106:49246;<wbr>transport=ws</a> SIP/2.0<br>
Record-Route:
<a class="m_-865090424726876519moz-txt-link-rfc2396E"><sip:10.0.1.200:5059;ftag=<wbr>d5de999de446df5165d773dac1f369<wbr>ec;lr=on></a><br>
Contact: "Megalokonomos A." <a class="m_-865090424726876519moz-txt-link-rfc2396E"><sip:694@10.0.1.200:45698></a><br>
User-Agent: OxO_SPG_103/012.001<br>
Content-Type: application/sdp<br>
To: <a class="m_-865090424726876519moz-txt-link-freetext">sip:694@10.0.1.200;tag=<wbr>4em4m1ah9r</a><br>
From: "Megalokonomos A."
<a class="m_-865090424726876519moz-txt-link-rfc2396E"><sip:610@10.0.1.200></a>;tag=<wbr>d5de999de446df5165d773dac1f369<wbr>ec<br>
Call-ID: <a class="m_-865090424726876519moz-txt-link-abbreviated" href="mailto:af3cc9085db1c8dd86050eb91d747249@10.0.1.200" target="_blank">af3cc9085db1c8dd86050eb91d7472<wbr>49@10.0.1.200</a><br>
CSeq: 659214613 ACK<br>
Via: SIP/2.0/UDP
10.0.1.200:5059;branch=<wbr>z9hG4bKf3de.<wbr>2fc1fc65cece765af47f9baf8bf090<wbr>6e.0;i=c<br>
Via: SIP/2.0/TCP
10.0.1.200:5080;rport=45698;<wbr>branch=<wbr>z9hG4bK89fca3417cd4e227b431514<wbr>5d96657c7<br>
Max-Forwards: 69<br>
Content-Length: 2960<br>
<br></span>
v=0<br>
o=default 14<br>
.....<br>
<br>
<br>
As OpenSIPS does not find the Route (former Record-Route) it
inserted into the dialog, the routing logic in the script does not
work as expected. According to RFC3261, the RR headers MUST be
mirrored back in 2xx replies.<br>
<br>
Let's try to hack to cope with the broken SIP stack on</tt><tt><tt>
OmniPCX. In script you have something like:<br>
<br>
</tt></tt>
<pre class="m_-865090424726876519escaped"> } else {
# ACK without matching transaction ->
# ignore and discard
exit;
}
Try replacing it with
} else {
# ACK without matching transaction ->
# ignore and discard
t_relay();
exit;
}
Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS.
Best regards,
</pre><span class="">
<pre class="m_-865090424726876519moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/<wbr>OpenSIPS_Bootcamp_2017.html</a>
</pre>
</span><div><div class="h5"><div class="m_-865090424726876519moz-cite-prefix">On 07/03/2017 01:08 PM, Bogdan-Andrei
Iancu wrote:<br>
</div>
<blockquote type="cite">
<tt>Hi Alex,<br>
<br>
As suspected, the ACK is not properly routed - see the
retransmissions of the 200OK + ACK. SImply based on the logs I
cannot see what the problem is - probably some missing
fix_nated_contact() for the replies coming from the WS party.<br>
<br>
Please make a pcap capture + opensips log (level 4) and send
them to me *offlist* !<br>
<br>
Best regards,<br>
</tt>
<pre class="m_-865090424726876519moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/<wbr>OpenSIPS_Bootcamp_2017.html</a>
</pre>
<div class="m_-865090424726876519moz-cite-prefix">On 06/30/2017 05:37 PM, Alex
Megalokonomos wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr"><img class="m_-865090424726876519mailtrack-img" style="float:right" alt="" src="data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7" height="0" width="0">I have attached the debug log so you
get a fuller picture. I hope that's ok
<div><br>
</div>
<div>(Incoming call to WS client 694 is the WS extension...610
is my normal desk phone which is connected to OmniPCX)
(10.0.1.63-> OpenSIPS ,10.0.1.200-> OmniPCX)
<div><br>
</div>
<div><br>
</div>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Fri, Jun 30, 2017 at 5:20 PM,
Bogdan-Andrei Iancu <span dir="ltr"><<a class="m_-865090424726876519moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank"></a><a class="m_-865090424726876519moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Good, there is
some progress :).<br>
<br>
On the incoming calls, if the WS get's the call, we
can park the part with the auth (it seems your
opensips script is accepting calls from unknown
sources...we can address this security hole later.<br>
<br>
So, if a call drop after 30 secs it usually means
there is no ACK. Can you make a mgrep capture on
OpenSIPS to grab the whole call flow ? (grab 5060 and
80 ports)<br>
<br>
Regards,<br>
</tt><span>
<pre class="m_-865090424726876519m_2726667119532378167moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519m_2726667119532378167moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519m_2726667119532378167moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/O<wbr>penSIPS_Bootcamp_2017.html</a>
</pre>
</span><div><div class="m_-865090424726876519h5"><div class="m_-865090424726876519m_2726667119532378167moz-cite-prefix">On 06/30/2017 04:52 PM, Alex
Megalokonomos wrote:
</div>
<blockquote type="cite">
<div dir="ltr"><img class="m_-865090424726876519m_2726667119532378167mailtrack-img" style="float:right" alt="" height="0" width="0">I think I set up uac_registrant
correctly.
<div>
</div>
<div>I can dial out from a ws client and the ws extension rings
from outside calls.</div>
<div>
</div>
<div>However:
a) on incoming calls, when ws client accepts, there is no
sound and the line is dropped after 30 secs or so</div>
<div>b) on outgoing calls, when the called extension accepts the
ws client immediately responds with 401 Unauthorised and then
BYE </div>
<div>
</div>
<div>b) I believe is what you mentioned here "<tt><tt>In
OpenSIPS, when receiving calls, you need to authorize (by
IP) the calls from </tt></tt><tt><tt><tt><tt>OmniPCX "</tt></tt></tt></tt>
</div>
<div><tt><tt><tt><tt>
</tt></tt></tt></tt></div>
<div><tt><tt><tt><tt>How do I do this?</tt></tt></tt></tt></div>
<div><tt><tt><tt><tt>
</tt></tt></tt></tt></div>
<div><tt><tt><tt><tt>and a) seems to be rtp proxy related since
I see the following errors in the logs¨</tt></tt></tt></tt></div>
<div>
</div>
<div><tt><tt><tt><tt><span>ERROR:rtpengine:rtpe_function_<wbr>call:
proxy replied with error: Unknown call-id</span>
</tt></tt></tt></tt></div>
<div><tt><tt><tt><tt><span>
</span></tt></tt></tt></tt></div>
<div><tt><tt><tt><tt><span>and</span></tt></tt></tt></tt></div>
<div><tt><tt><tt><tt><span>
</span></tt></tt></tt></tt></div>
<div><tt><tt><tt><tt><span>no matching transaction</span></tt></tt></tt></tt></div>
</div>
<div class="gmail_extra">
<div class="gmail_quote">On Fri, Jun 30, 2017 at 2:27 PM,
Bogdan-Andrei Iancu <span dir="ltr"><<a class="m_-865090424726876519m_2726667119532378167moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>I checked the
script you mentioned and it does not help you - it has
only UDP (no WS), it is really basic and it does not
handle any REGISTER stuff, which is the trickiest - see
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-freetext" href="https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/" target="_blank">https://blog.opensips.org/2016<wbr>/12/13/how-to-proxy-sip-regist<wbr>rations/</a>
or
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-freetext" href="https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/" target="_blank">https://blog.opensips.org/2016<wbr>/12/20/mid-registrar-scalable-<wbr>registration-and-call-forking/</a>
Maybe you can start with handling REGISTERs - what you
need (on top of the script from the WSS tutorial) is to
add this uac_registrant, to have the WS extensions
registered into </tt><tt>OmniPCX with a contact URI
pointing back to OpenSIPS IP:
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/uac_registrant.html" target="_blank">http://www.opensips.org/html/d<wbr>ocs/modules/2.3.x/uac_registra<wbr>nt.html</a>
Let me know if you get stuck in this first step.
Regards,
</tt><span>
<pre class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/O<wbr>penSIPS_Bootcamp_2017.html</a>
</pre>
</span><div><div class="m_-865090424726876519m_2726667119532378167h5"><div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-cite-prefix">On 06/30/2017 12:22 PM, Alex
Megalokonomos wrote:
</div>
<blockquote type="cite">
<div dir="ltr"><img class="m_-865090424726876519m_2726667119532378167m_-701523849835307735mailtrack-img" style="float:right" alt="" height="0" width="0">Hello Bogdan,
<div>
</div>
<div>First of all, thanks for your time.</div>
<div>
</div>
<div>Unfortunately my SIP/OpensSIPS skills are what I've managed
to learn in the last couple of days. I am a programmer but
I've never had to work on SIP stuff before.</div>
<div>
</div>
<div>Frankly to me, both solutions sound equally difficult since
I have no idea where to start. (And to be honest, I expected
the first to be simpler)</div>
<div>
</div>
<div>I found this <a href="https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/" target="_blank">https://blog.voipxswitch.<wbr>com/2015/03/27/kamailio-basic-<wbr>sip-proxy-all-requests-setup/</a>
and tried to port the config to OpenSIPS since from what I
understand Kamailio and OpenSIPS share a common codebase to an
extent but was unsuccesful.
</div>
<div>
</div>
<div>In your second scenario, I am not interested in WS->WS
calls so that auth part is not an issue.</div>
<div>
</div>
<div>So I guess I need the uac_registrar, authorize by IP and
usrloc parts.</div>
<div>
</div>
<div>Any relevant documentation to get me started since I'm
still not clear on what I need to change?</div>
<div>
</div>
<div>Best regards,</div>
<div>Alex</div>
</div>
<div class="gmail_extra">
<div class="gmail_quote">On Fri, Jun 30, 2017 at 11:29 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Hi Alex,
To make a kind of WS<>UDP gateway you need a
complete rework of the script presented in the tutorial,
as it is a completely different SIP scenario. Not sure
what are your SIP/OpenSIPS skills.
But, there is a simpler alternative . Instead of a GW,
you can make OpenSIPS as a sub-server for the WS
extensions:
Registration handling:
1) WS extensions register only with OpenSIPS (as right
now) - authentication is done by OpenSIPS
2) OpenSIPS registers the 3 extensions into </tt><tt>OmniPCX
using the uac_registrar
By this, we simply add the uac_registration and you
achieve kind of decoupled 2 steps registration (with a
minimum change in the cfg)
Inbound calls:
1) </tt><tt><tt>OmniPCX will send all the calls (from
other extensions) for the WS extension to OpenSIPS
(due the registration via uac_registrar) - this is
default behavior , so nothing to change
2) In OpenSIPS, when receiving calls, you need to
authorize (by IP) the calls from </tt></tt><tt><tt><tt><tt>OmniPCX
- and as the current script does, you will handle
them via the local opensips usrloc -> calls are
sent to WS extension
</tt></tt></tt></tt>
<tt><tt><tt><tt><tt>Outbound calls:
1) when you receive a call from a WS extension,
you have to check if the call is for a local
extension (on opensips) or for an extension in </tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt>OmniPCX
2) if call is local (WS to WS) you will do
authentication for the call
3) if the call is to be sent to </tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX,
simply send the call to </tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX
without auth - the
auth will be done by </tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt><tt>OmniPCX
as for any
other
extension
Hopefully this
will work for
you :)
</tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt></tt>Best
regards,
</tt><span>
<pre class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/O<wbr>penSIPS_Bootcamp_2017.html</a>
</pre>
</span><div><div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735h5"><div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565moz-cite-prefix">On 06/29/2017 11:54 AM, Alex
Megalokonomos wrote:
</div>
<blockquote type="cite">
<div dir="ltr"><img class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565mailtrack-img" style="float:right" alt="" height="0" width="0">Hello Bogdan,
<div>
</div>
<div>Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP
in order for it to work) is exactly what we're looking for.</div>
<div>
</div>
<div>Unfortunately our Alcatel OmniPCX call center is a
proprietary system that only allows for a limited number of
SIP extensions (served from what appears to be an outdated
customised Kamailio 3.2.2 from what I can tell from the
headers.</div>
<div>
</div>
<div>For our normal internal office use it all works fine.</div>
<div>
</div>
<div>However we have 3 customer support lines that are currently
routed to 3 extensions via OmniPCX.</div>
<div>
</div>
<div>We want to integrate these to our custom web-based CRM and
the best way for us to do it is to use something like SIP js
to handle and log calls, identify calling parties, bring up
customer details etc.</div>
<div>
</div>
<div>Since the kamailio version inside OmniPCX does not support
ws/webrtc we are looking to set up Opensips in exactly the way
you described as a gateway/proxy for everything in order to
convert the UDP-only sip extensions to ws+ webRTC capable
ones.</div>
<div>
</div>
<div>I have used this tutorial <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank">http://www.opensips.o<wbr>rg/Documentation/Tutorials-Web<wbr>Socket-2-1</a>
to get what I assume is half the work (for RTP proxying) but
I havent figured out the rest yet.</div>
<div>
</div>
<div>Best regards,</div>
<div>Alex</div>
</div>
<div class="gmail_extra">
<div class="gmail_quote">On Thu, Jun 29, 2017 at 11:43 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> <tt>Hi Alex,
First, some questions regarding the desired topology:
1) the WS end-points should register in OpenSIPS or
all the way into Kamailio ?
2) also, the calls from the WS end-points should be
all the time sent to Kamailio ?
More or less, what I'm asking is : is OpenSIPS suppose
to act as a gateway from WS to UDP , but pass all the
resulting traffic to Kamailio ?
Regards,
</tt>
<pre class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/O<wbr>penSIPS_Bootcamp_2017.html</a>
</pre><div><div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565h5">
<div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-cite-prefix">On 06/28/2017 12:47 PM, Alex
Megalokonomos wrote:
</div>
</div></div><blockquote type="cite"><div><div class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565h5">
<div dir="ltr"><img class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616mailtrack-img" style="float:right" alt="" height="0" width="0">Hello,
<div>
</div>
<div>We have the following scenario: our office call center is
an Alcatel OmniPCX Office setup.</div>
<div>
</div>
<div>This handles most of our needs and also provides 4 SIP
extensions.</div>
<div>
</div>
<div>These are provided by what appears to be a Kamailio SIP
server v 3.2.2 (no webrtc or websockets support)</div>
<div>
</div>
<div>What we would like to do is set up an OpenSIPS instance to
handle WebRTC and proxy everything to this Kamailio SIP
server.</div>
<div>
</div>
<div>The idea is to allow a web client (using sip js or
something similar) to register / make / receive calls as one
of the Kamailio extensions.</div>
<div>
</div>
<div>
<div>I think half of the configuration is this : <a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank">http://www.opensips.org/Docu<wbr>mentation/Tutorials-WebSocket-<wbr>2-1</a></div>
<div>
</div>
<div>which I've already completed and indeed, clients can
register to opensips and chat/make calls over websockets
between them.</div>
<div>
</div>
<div>How do I go about proxying registrations/invites/etc to
the kamailio server instead?</div>
<div>
</div>
<div>best regards</div>
</div>
</div>
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</div></div><span><pre>______________________________<wbr>_________________
Users mailing list
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a class="m_-865090424726876519m_2726667119532378167m_-701523849835307735m_-1652678207613874565m_-6874691586421181616moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-<wbr>bin/mailman/listinfo/users</a>
</pre>
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<pre>______________________________<wbr>_________________
Users mailing list
<a class="m_-865090424726876519moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org" target="_blank">Users@lists.opensips.org</a>
<a class="m_-865090424726876519moz-txt-link-freetext" href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-<wbr>bin/mailman/listinfo/users</a>
</pre>
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