25 Aug
2016
25 Aug
'16
3:38 p.m.
I have configured Kamailio to pass a registration request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to the
VoIP provider but I'd like to have inbound calls working as well. I see the
inbound invites coming into the Kamailio server but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to fix this problem? Below is
my config.
Thank you,
Mike
#!KAMAILIO
#
# Edge proxy configuration
#
#!subst "/REGISTRAR_IP/xxxxxxxxxx/" #IP address of voip provider
#!subst "/REGISTRAR_PORT/5060/"
#!substdef "/FLOW_TIMER/20/"
debug=2
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=4
alias="example.com"
#mpath="/usr/lib64/kamailio/modules"
mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules"
tcp_connection_lifetime=30 # FLOW_TIMER + 10
auto_aliases=no
mhomed=0
listen=udp:X.X.X.X:531 advertise X.X.X.X:531
alias="X.X.X.X:531"
alias=X.X.X.X:531
alias="X.X.X.X:531"
alias=X.X.X.X:531
port=531
####### Modules Section ########
loadmodule "tm.so"
loadmodule "sl.so"
#loadmodule "outbound.so"
loadmodule "rr.so"
loadmodule "path.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "mi_rpc.so"
loadmodule "mi_fifo.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "rtimer.so"
loadmodule "sqlops.so"
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- tm params -----
modparam("tm", "failure_reply_mode", 3)
# ----- rr params -----
modparam("rr", "append_fromtag", 0)
####### Routing Logic ########
request_route {
route(REQINIT);
if (is_method("CANCEL")) {
if (t_check_trans()) {
route(RELAY);
}
exit;
}
route(WITHINDLG);
t_check_trans();
if (is_method("REGISTER")) {
remove_hf("Route");
add_path();
$du = "sip:REGISTRAR_IP:REGISTRAR_PORT";
} else {
if (is_method("INVITE|SUBSCRIBE"))
record_route();
if ($si == "REGISTRAR_IP" && $sp ==
"REGISTRAR_PORT") {
if (!loose_route()) {
switch($rc) {
case -2:
sl_send_reply("403", "Forbidden");
exit;
default:
sl_reply_error();
exit;
}
}
t_on_failure("FAIL_OUTBOUND");
} else {
if ($rU == $null) {
sl_send_reply("484", "Address Incomplete");
exit;
}
remove_hf("Route");
$du = "sip:REGISTRAR_IP:REGISTRAR_PORT";
}
}
route(RELAY);
}
route[RELAY] {
if (!t_relay()) {
sl_reply_error();
}
exit;
}
route[REQINIT] {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
route[WITHINDLG] {
if (has_totag()) {
if (!loose_route()) {
switch($rc) {
case -2:
sl_send_reply("403", "Forbidden");
exit;
default:
if (is_method("ACK")) {
if ( t_check_trans() ) {
route(RELAY);
exit;
} else {
exit;
}
}
sl_send_reply("404","Not Found");
}
} else {
if (is_method("NOTIFY")) {
record_route();
}
route(RELAY);
}
exit;
}
}
onreply_route {
if (!t_check_trans()) {
drop;
}
if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299)
{
remove_hf("Flow-Timer");
if ($(hdr(Require)[*])=~"outbound")
insert_hf("Flow-Timer: FLOW_TIMER\r\n",
"Call-ID");
}
}
failure_route[FAIL_OUTBOUND]{
if (t_branch_timeout() || !t_branch_replied()) {
send_reply("430", "Flow Failed");
}
}
25 Aug
25 Aug
3:57 p.m.
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike Patterson wrote:
I have configured Kamailio to pass a registration
request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to the
VoIP provider but I'd like to have inbound calls working as well. I see the
inbound invites coming into the Kamailio server but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to fix this problem? Below is
my config.
You should look into the NAT handling tricks:
http://www.kamailio.org/docs/modules/stable/modules/nathelper.html#nathelpe…
and
http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.f.…
Take a look at the default config NATMANAGE routes and enable "natting"
for all sources except you upstream server.
Alternatively if the registrar of the voip provider supports Path,
enable that in kamailio use
http://kamailio.org/docs/modules/stable/modules/path.html#idp38122172
in the registers to add the header and set
http://kamailio.org/docs/modules/stable/modules/path.html#idp38033932
to handle the INVITEs
4:19 p.m.
Hi Matt,
Check out this post and config:
https://blog.voipxswitch.com/2016/08/11/kamailio-and-freeswitch-on-the-same…
It demonstrates how to handle NAT & forward a REGISTER to FreeSWITCH (on
localhost)
Thanks,
Emmanuel
On Thu, Aug 25, 2016 at 9:57 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike
Patterson wrote:
I have configured Kamailio to pass a registration
request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to
the
VoIP provider but I'd like to have inbound
calls working as well. I see
the
inbound invites coming into the Kamailio server
but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to
fix this problem?
Below is
my config.
You should look into the NAT handling tricks:
http://www.kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.set_contact_alias
and
http://kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.f.handle_ruri_alias
Take a look at the default config NATMANAGE routes and enable "natting"
for all sources except you upstream server.
Alternatively if the registrar of the voip provider supports Path,
enable that in kamailio use
http://kamailio.org/docs/modules/stable/modules/path.html#idp38122172
in the registers to add the header and set
http://kamailio.org/docs/modules/stable/modules/path.html#idp38033932
to handle the INVITEs
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
2 Sep
2 Sep
7:13 p.m.
Am 25.08.2016 um 16:19 schrieb E. Schmidbauer:
Hi Matt,
Check out this post and config:
https://blog.voipxswitch.com/2016/08/11/kamailio-and-freeswitch-on-the-same…
It demonstrates how to handle NAT & forward a REGISTER to FreeSWITCH
(on localhost)
Thanks,
Emmanuel
Just a comment (I found this thread as I currently have the some
problem). Both provided solutions do use PATH which of course requires
PATH support on the registrar. If the reigstrar does not support PATH
then things get complicated.
I see 2 solutions:
a) if the registrar does NAT traversal (and thus always send SIP
messages to the IP:port it received the REGISTER), does not manipulate
the received Contact header and the client does not care about
manipulated Contact header it may be possible by just applying
"fix_nated_contact()" and route messages from the registrar to the
client by using the R-URI in the incoming request.
b) if the Contact header must be preserved (strict clients) or if the
registrar does not apply NAT traversal, then it is necessary to do heavy
message rewriting and registration in the proxy too:
- the URI in the Contact forwarded to the registrar must point to the proxy
- this URI must be mapped by the proxy to the clients received/NAT address
- depending on the requirements (e.g. multiple registrars, multiple
domains) the mapping can be simple or complex
Thus, I think a generic solution for registrars without PATH support
involves plenty of message manipulation and lot of time and testing.
regards
Klaus
5 Sep
5 Sep
10:07 a.m.
On Fri, Sep 02, 2016 at 07:13:13PM +0200, Klaus Darilion wrote:
I see 2 solutions:
a) fix_nated_contact
...
b) if the Contact header must be preserved (strict
clients) or if the
registrar does not apply NAT traversal, then it is necessary to do heavy
message rewriting and registration in the proxy too:
- the URI in the Contact forwarded to the registrar must point to the proxy
- this URI must be mapped by the proxy to the clients received/NAT address
- depending on the requirements (e.g. multiple registrars, multiple domains)
the mapping can be simple or complex
Another solution might be to use the topoh/topos modules. Haven't ever
tried it on a machine that forwards REGISTERs having glanced the topos
code it should work for any SIP message.
25 Aug
25 Aug
4:21 p.m.
Err sorry Mike (not matt)
On Thu, Aug 25, 2016 at 9:57 AM, Daniel Tryba <d.tryba(a)pocos.nl> wrote:
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike
Patterson wrote:
I have configured Kamailio to pass a registration
request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to
the
VoIP provider but I'd like to have inbound
calls working as well. I see
the
inbound invites coming into the Kamailio server
but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to
fix this problem?
Below is
my config.
You should look into the NAT handling tricks:
http://www.kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.set_contact_alias
and
http://kamailio.org/docs/modules/stable/modules/
nathelper.html#nathelper.f.handle_ruri_alias
Take a look at the default config NATMANAGE routes and enable "natting"
for all sources except you upstream server.
Alternatively if the registrar of the voip provider supports Path,
enable that in kamailio use
http://kamailio.org/docs/modules/stable/modules/path.html#idp38122172
in the registers to add the header and set
http://kamailio.org/docs/modules/stable/modules/path.html#idp38033932
to handle the INVITEs
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:34 p.m.
Thank you. I am reviewing the doc. I am new to Kamailio so it will take some time for
me.
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of E. Schmidbauer
Sent: Thursday, August 25, 2016 10:22 AM
To: d.tryba(a)pocos.nl; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio - passing registration to another sip server
Err sorry Mike (not matt)
On Thu, Aug 25, 2016 at 9:57 AM, Daniel Tryba <d.tryba(a)pocos.nl
<mailto:d.tryba@pocos.nl> > wrote:
On Thu, Aug 25, 2016 at 09:38:12AM -0400, Mike Patterson wrote:
I have configured Kamailio to pass a registration
request to another VoIP
provider. The reason I am doing this is to provide a sip port for users
where the ISP is blocking SIP. I am able to pass the registrations to the
VoIP provider but I'd like to have inbound calls working as well. I see the
inbound invites coming into the Kamailio server but it appears that the
server does not 'know' about the UA that is registered 'through' the
server
to the VoIP provider. Can anyone tell me how to fix this problem? Below is
my config.
You should look into the NAT handling tricks:
http://www.kamailio.org/docs/modules/stable/modules/nathelper.html#nathelpe…
and
http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.f.…
Take a look at the default config NATMANAGE routes and enable "natting"
for all sources except you upstream server.
Alternatively if the registrar of the voip provider supports Path,
enable that in kamailio use
http://kamailio.org/docs/modules/stable/modules/path.html#idp38122172
in the registers to add the header and set
http://kamailio.org/docs/modules/stable/modules/path.html#idp38033932
to handle the INVITEs
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
26 Aug
26 Aug
10:51 a.m.
On Thu, Aug 25, 2016 at 01:34:30PM -0400, Mike Patterson wrote:
Thank you. I am reviewing the doc. I am new to
Kamailio so it will
take some time for me.
Good luck then :)
Here is my slightly redacted config for a setup similar to yours (which
is just a loadbalancer/proxy), which uses both contact rewriting
(invites) and Path (registers) and dispatcher to contact the backends.
It is based on the default config supplied with kamailio (so look there
for REQINIT and WITHINDLG and anything not listed here) and has NATting
stuff enabled by default (for SIP, RTP is handled elsewhere). I don't
pretend to know that it works perfectly, it just works fine for me.
request_route {
route(REQINIT);
route(NATDETECT);
route(WITHINDLG);
if (is_method("INVITE|SUBSCRIBE"))
{
record_route();
}
if(!is_in_subnet($si, "x.x.x.x/30"))
{
add_path_received("fromlb");
ds_select_dst(1, 0);
t_on_failure("RTF_DISPATCH");
}
else
{
remove_hf("Route");
loose_route();
}
route(RELAY);
}
route[NATDETECT] {
if(!is_in_subnet($si, "x.x.x.x/30"))
{
if (!is_method("REGISTER") && is_first_hop())
{
set_contact_alias();
}
}
return;
}
route[NATMANAGE] {
if(!is_in_subnet($si, "x.x.x.x/30"))
{
if (is_reply() && is_first_hop())
{
set_contact_alias();
}
}
return;
}
1:39 p.m.
Thank you for your response. Can you tell me what value I need to put in is_in_subnet?
Is this the local network of the Kamailio server or is it the network of the down-stream
sip server?
Do you of anyone who might be willing to work with me on this configuration for a fee or
bounty?
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Tryba
Sent: Friday, August 26, 2016 4:52 AM
To: sr-users(a)lists.sip-router.org
Subject: Re: [SR-Users] Kamailio - passing registration to another sip server
On Thu, Aug 25, 2016 at 01:34:30PM -0400, Mike Patterson wrote:
Thank you. I am reviewing the doc. I am new to
Kamailio so it will
take some time for me.
Good luck then :)
Here is my slightly redacted config for a setup similar to yours (which is just a
loadbalancer/proxy), which uses both contact rewriting
(invites) and Path (registers) and dispatcher to contact the backends.
It is based on the default config supplied with kamailio (so look there for REQINIT and
WITHINDLG and anything not listed here) and has NATting stuff enabled by default (for SIP,
RTP is handled elsewhere). I don't pretend to know that it works perfectly, it just
works fine for me.
request_route {
route(REQINIT);
route(NATDETECT);
route(WITHINDLG);
if (is_method("INVITE|SUBSCRIBE"))
{
record_route();
}
if(!is_in_subnet($si, "x.x.x.x/30"))
{
add_path_received("fromlb");
ds_select_dst(1, 0);
t_on_failure("RTF_DISPATCH");
}
else
{
remove_hf("Route");
loose_route();
}
route(RELAY);
}
route[NATDETECT] {
if(!is_in_subnet($si, "x.x.x.x/30"))
{
if (!is_method("REGISTER") && is_first_hop())
{
set_contact_alias();
}
}
return;
}
route[NATMANAGE] {
if(!is_in_subnet($si, "x.x.x.x/30"))
{
if (is_reply() && is_first_hop())
{
set_contact_alias();
}
}
return;
}
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
6:16 p.m.
On Fri, Aug 26, 2016 at 07:39:02AM -0400, Mike Patterson wrote:
Thank you for your response. Can you tell me what
value I need to put
in is_in_subnet? Is this the local network of the Kamailio server or
is it the network of the down-stream sip server?
It is the subnet (or ip) of your downstream sip server.
Do you of anyone who might be willing to work with me
on this
configuration for a fee or bounty?
I can only speak for myself: I try to answer questions to help others
solve problems that I have figured out myself (many times by digging in
the archives here) and to see if there are better solutions available.
I'm not willing nor do I have the time to help commercially myself
directly. Just ask your questions here and hope someone will reply, the
only alternative is to hire a Kamailio consultant:
https://www.kamailio.org/w/business-directory/
Or try the commercial mailinglist: business(a)lists.kamailio.org
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9 comments
4 participants
participants (4)
-
Daniel Tryba -
E. Schmidbauer -
Klaus Darilion -
Mike Patterson