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List: serusers
Subject: Re: [SR-Users] relay sip-tls vs sips-tcp
From: Henning Westerholt <hw () gilawa ! com>
Date: 2022-10-21 9:49:06
Message-ID: AM6PR05MB54097A231EEBEF1C109ACD9FBF2D9 () AM6PR05MB5409 ! eurprd05 ! prod ! outlook ! com
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Hello,
the „sips" URI scheme is not really useful nowadays and IMHO should not be used. \
You probably find in the archives some old discussions about it.
Use "sip" and "transport=tls" if you want to have a secure connection.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-users <sr-users-bounces@lists.kamailio.org> On Behalf Of beer Ll
Sent: Thursday, October 20, 2022 9:22 AM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] relay sip-tls vs sips-tcp
Hi James
Thanks , You confirm my idea .
I will write to the UAC developers and check this problem asking to fix it.
Is safer use sip-tls or sips-tcp ?
Best Regards
Leo
On Wed, Oct 19, 2022 at 2:07 PM James Browne \
<james@frideo.com<mailto:james@frideo.com>> wrote: You make calls using SIP over TLS \
and it's OK. You make calls using SIPS and it's not OK.
The 200-OK Contact is this
- Contact: <sip:172.16.0.2:5060<http://172.16.0.2:5060>>
The RURI in the ACK is this.
- sips:172.16.0.2:5060;transport=tcp SIP/2.0
The client should be using _exactly_ the same URI in the ACK as was in the Contact in \
the 200-OK response. The client is getting it wrong \
(https://datatracker.ietf.org/doc/html/rfc3261#section-12.2.1.1).
With the ACK, the Routeset is this.
- ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0
- Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
- Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
This is asking kamailio to send the ACK to 172.16.0.2 over TLS on port 5060 (from the \
RURI), but use a UDP socket to do it (the second Route header field). This can't \
work. The client should be fixed. (Else you may try getting asterisk to use sips, but \
maybe that's not going to be easy.)
James
On Thu, 13 Oct 2022 at 06:30, beer Ll \
<llcfhllml@gmail.com<mailto:llcfhllml@gmail.com>> wrote: Hi everyone,
I'm using Kamailio as TLS gateway/filter for an internal Asterisk server
the network schema is :
UAC (tls) --- INTERNET --- (tls) KAMAILIO (sip udp) --- LAN --- (sip udp) ASTERISK
with kamailio in multi-homed mode
WAN network interface for sip tls
LAN network interface for sip udp to asterisk server
UAC address 80.0.0.1
KAMAILIO Wan address 80.0.0.2
KAMAILIO Lan address 172.16.0.2
ASTERISK Lan address 172.16.0.3
SIP-TLS call example
If the UAC use tls(sip) all works good
[cid:image001.jpg@01D8E543.1E954C20]
SIPS call example
If the same UAC use his default settings tls(sips) , there are problems with ACK and \
BYE packet
[cid:image002.jpg@01D8E543.1E954C20]
the SIP OK SDP packet from kamailio to UAC is
2022/10/10 09:28:47.854721 80.0.0.2:5061<http://80.0.0.2:5061> -> \
80.0.0.1:49992<http://80.0.0.1:49992> SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.0.1:49992;rport=49992;received=80.0.0.1;branch=z9hG4bKM01j360VrBdH5VSV
Record-Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Record-Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7
From: <sips:200@pbx.voip.com<mailto:sips%3A200@pbx.voip.com>>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F
To: <sips:*43@pbx.voip.com<mailto:43@pbx.voip.com>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96
CSeq: 2 INVITE
Contact: <sip:172.16.0.2:5060<http://172.16.0.2:5060>>
P-Asserted-Identity: "xxxxxxxxx" <sips:*43@pbx.voip.com<mailto:43@pbx.voip.com>>
Content-Type: application/sdp
and the UAC send the ACK and BYE from a different tcp port and to: \
sips:172.16.0.2:5060;transport=tcp
2022/10/10 09:28:48.495365 80.0.0.1:49996<http://80.0.0.1:49996> -> \
80.0.0.2:5061<http://80.0.0.2:5061> ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TLS 192.168.0.1:49996;branch=z9hG4bKppftdQze20lnwT41;rport
Route: <sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F>
Max-Forwards: 70
To: <sips:*43@pbx.voip.com<mailto:43@pbx.voip.com>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96
From: <sips:200@pbx.voip.com<mailto:sips%3A200@pbx.voip.com>>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7
CSeq: 2 ACK
kamailio error log
WARNING: <core> [core/forward.c:229]: get_send_socket2(): protocol/port mismatch \
(forced udp:172.16.0.2:5060<http://172.16.0.2:5060>, to \
tls:172.16.0.3:5060<http://172.16.0.3:5060>)
How can I solve this ?
Best Regards
Leo
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Important: keep the mailing list in the recipients, do not reply only to the sender!
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<p class="MsoNormal"><span \
style="mso-fareast-language:EN-US">Hello,<o:p></o:p></span></p> <p \
class="MsoNormal"><span \
style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">the „sips" \
URI scheme is not really useful nowadays and IMHO should not be used. You probably \
find in the archives some old discussions about it.<o:p></o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Use "sip" and \
"transport=tls" if you want to have a secure connection.<o:p></o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US">Cheers,<o:p></o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US">Henning<o:p></o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">-- <o:p> \
</o:p></span></p> <p class="MsoNormal"><span lang="EN-GB" \
style="mso-fareast-language:EN-US">Henning Westerholt – </span><span \
style="mso-fareast-language:EN-US"><a href="https://skalatan.de/blog/"><span \
lang="EN-GB" style="color:#0563C1">https://skalatan.de/blog/</span></a></span><span \
lang="EN-GB" style="mso-fareast-language:EN-US"><o:p></o:p></span></p> <p \
class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Kamailio \
services – </span><span style="mso-fareast-language:EN-US"><a \
href="https://gilawa.com/"><span lang="EN-GB" \
style="color:#0563C1">https://gilawa.com</span></a></span><span lang="EN-GB" \
style="mso-fareast-language:EN-US"><o:p></o:p></span></p> <p class="MsoNormal"><span \
lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p> <div \
style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm"> <p \
class="MsoNormal" style="margin-left:35.4pt"><b>From:</b> sr-users \
<sr-users-bounces@lists.kamailio.org> <b>On Behalf Of </b>beer Ll<br>
<b>Sent:</b> Thursday, October 20, 2022 9:22 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List \
<sr-users@lists.kamailio.org><br> <b>Subject:</b> Re: [SR-Users] relay sip-tls \
vs sips-tcp<o:p></o:p></p> </div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Hi James<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Thanks , You confirm my idea \
.<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">I will write to the UAC developers \
and check this problem asking to fix it.<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Is safer use sip-tls or sips-tcp \
?<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Best Regards<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Leo<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">On Wed, Oct 19, 2022 at 2:07 PM James \
Browne <<a href="mailto:james@frideo.com">james@frideo.com</a>> \
wrote:<o:p></o:p></p> </div>
<blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm \
6.0pt;margin-left:4.8pt;margin-right:0cm"> <div>
<p class="MsoNormal" style="margin-left:35.4pt">You make calls using SIP over TLS and \
it's OK.<br> You make calls using SIPS and it's not OK.<br>
<br>
The 200-OK Contact is this<br>
- Contact: <sip:<a href="http://172.16.0.2:5060" \
target="_blank">172.16.0.2:5060</a>><br> <br>
The RURI in the ACK is this.<br>
- <a href="sips:172.16.0.2:5060;transport=tcp">sips:172.16.0.2:5060;transport=tcp</a> \
SIP/2.0<br> <br>
The client should be using _exactly_ the same URI in the ACK as was in the Contact in \
the 200-OK response. The client is getting it wrong (<a \
href="https://datatracker.ietf.org/doc/html/rfc3261#section-12.2.1.1" \
target="_blank">https://datatracker.ietf.org/doc/html/rfc3261#section-12.2.1.1</a>).<br>
<br>
With the ACK, the Routeset is this.<br>
- ACK <a href="sips:172.16.0.2:5060;transport=tcp">sips:172.16.0.2:5060;transport=tcp</a> \
SIP/2.0<br>
- Route: <<a href="sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA \
89D3BDFE0C8C8F">sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br>
- Route: <<a href="sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C \
8F">sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br> \
<br> This is asking kamailio to send the ACK to 172.16.0.2 over TLS on port 5060 \
(from the RURI), but use a UDP socket to do it (the second Route header field). This \
can't work. The client should be fixed.<o:p></o:p></p> <div>
<p class="MsoNormal" style="margin-left:35.4pt">(Else you may try getting asterisk to \
use sips, but maybe that's not going to be easy.)<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">James<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">On Thu, 13 Oct 2022 at 06:30, beer Ll \
<<a href="mailto:llcfhllml@gmail.com" target="_blank">llcfhllml@gmail.com</a>> \
wrote:<o:p></o:p></p> </div>
<blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm \
6.0pt;margin-left:4.8pt;margin-right:0cm"> <div>
<p class="MsoNormal" style="margin-left:35.4pt">Hi everyone, <o:p></o:p></p>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">I'm using Kamailio as TLS \
gateway/filter for an internal Asterisk server<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">the network schema is : \
<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">UAC (tls) --- INTERNET --- \
(tls) KAMAILIO (sip udp) --- LAN --- (sip udp) ASTERISK<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"> <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">with kamailio in multi-homed mode
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">WAN network interface for sip tls
<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">LAN network interface for sip \
udp to asterisk server <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">UAC address 80.0.0.1<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">KAMAILIO Wan address \
80.0.0.2<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">KAMAILIO Lan address \
172.16.0.2<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">ASTERISK Lan address \
172.16.0.3<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" \
style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt">
<o:p> </o:p></p>
</div>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">SIP-TLS call example<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">If the UAC use tls(sip) all works \
good <o:p></o:p></p>
</div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><img border="0" width="558" \
height="363" style="width:5.8125in;height:3.7812in" id="Bild_x0020_1" \
src="cid:image001.jpg@01D8E543.1E954C20"><o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">SIPS call example<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">If the same UAC use his default \
settings tls(sips) , there are problems with ACK and BYE packet<o:p></o:p></p> \
</div> <div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><img border="0" width="559" \
height="367" style="width:5.8229in;height:3.8229in" id="Bild_x0020_2" \
src="cid:image002.jpg@01D8E543.1E954C20"><o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">the SIP OK SDP packet from kamailio \
to UAC is <o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" \
style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt">
2022/10/10 09:28:47.854721 <a href="http://80.0.0.2:5061" \
target="_blank">80.0.0.2:5061</a> -> <a href="http://80.0.0.1:49992" \
target="_blank">80.0.0.1:49992</a><br> SIP/2.0 200 OK<br>
Via: SIP/2.0/TLS 192.168.0.1:49992;rport=49992;received=80.0.0.1;branch=z9hG4bKM01j360VrBdH5VSV<br>
Record-Route: <<a \
href="sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F">sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br>
Record-Route: <<a \
href="sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F"> \
sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br>
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7<br>
From: <<a href="mailto:sips%3A200@pbx.voip.com" \
target="_blank">sips:200@pbx.voip.com</a>>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F<br>
To: <sips:*<a href="mailto:43@pbx.voip.com" \
target="_blank">43@pbx.voip.com</a>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96<br>
CSeq: 2 INVITE<br>
Contact: <sip:<a href="http://172.16.0.2:5060" \
target="_blank">172.16.0.2:5060</a>><br>
P-Asserted-Identity: "xxxxxxxxx" <sips:*<a href="mailto:43@pbx.voip.com" \
target="_blank">43@pbx.voip.com</a>><br>
Content-Type: application/sdp<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">and the UAC send the ACK and BYE from \
a different tcp port and to: <a \
href="sips:172.16.0.2:5060;transport=tcp">sips:172.16.0.2:5060;transport=tcp</a> \
<o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">2022/10/10 09:28:48.495365 <a \
href="http://80.0.0.1:49996" target="_blank"> 80.0.0.1:49996</a> -> <a \
href="http://80.0.0.2:5061" target="_blank">80.0.0.2:5061</a><br> ACK <a \
href="sips:172.16.0.2:5060;transport=tcp">sips:172.16.0.2:5060;transport=tcp</a> \
SIP/2.0<br>
Via: SIP/2.0/TLS 192.168.0.1:49996;branch=z9hG4bKppftdQze20lnwT41;rport<br>
Route: <<a href="sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89 \
D3BDFE0C8C8F">sip:80.0.0.2:5061;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br>
Route: <<a href="sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F">sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F</a>><br>
Max-Forwards: 70<br>
To: <sips:*<a href="mailto:43@pbx.voip.com" \
target="_blank">43@pbx.voip.com</a>>;tag=961d0e22-a4f0-453c-9870-6a41578afc96<br>
From: <<a href="mailto:sips%3A200@pbx.voip.com" \
target="_blank">sips:200@pbx.voip.com</a>>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F<br>
Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7<br>
CSeq: 2 ACK<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">kamailio error log<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">WARNING: <core> \
[core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced udp:<a \
href="http://172.16.0.2:5060" target="_blank">172.16.0.2:5060</a>, to tls:<a \
href="http://172.16.0.3:5060" target="_blank">172.16.0.3:5060</a>)<o:p></o:p></p> \
</div> <div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">How can I solve this ?<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Best Regards<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt">Leo<o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
</div>
<p class="MsoNormal" \
style="margin-left:35.4pt">__________________________________________________________<br>
Kamailio - Users Mailing List - Non Commercial Discussions<br>
<a href="mailto:sr-users@lists.kamailio.org" \
target="_blank">sr-users@lists.kamailio.org</a><br>
Important: keep the mailing list in the recipients, do not reply only to the \
sender!<br> Edit mailing list options or unsubscribe:<br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" \
target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
</blockquote>
</div>
<p class="MsoNormal" \
style="margin-left:35.4pt">__________________________________________________________<br>
Kamailio - Users Mailing List - Non Commercial Discussions<br>
<a href="mailto:sr-users@lists.kamailio.org" \
target="_blank">sr-users@lists.kamailio.org</a><br>
Important: keep the mailing list in the recipients, do not reply only to the \
sender!<br> Edit mailing list options or unsubscribe:<br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" \
target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
</blockquote>
</div>
</div>
</body>
</html>
["image001.jpg" (image/jpeg)]
["image002.jpg" (image/jpeg)]
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender!
Edit mailing list options or unsubscribe:
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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