Take a look at freeSWITCH
On Mon, 17 Jan 2022 at 00:58, Chad <[email protected]
<mailto:[email protected]>> wrote:
Hmm, it did not fix it (calls still work with my other carriers).
It looks to me like it should work, it does use the external IP for
everything.
It generates an error in the log about making your existing address:
topoh [topoh_mod.c:179]: mod_init(): mask address matches myself
[209.###.###.###]
Here is ther 200 and ACK.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bKf229.9bb425c2.0
Via: SIP/2.0/UDP
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bKrs8bqi00cg14535baf70.1
Record-Route:
<sip:209.###.###.###;line=sr-1RaGXxdGcxdGcxdGcxgTp8eVKxT-jxeE1xT-jxehH02vI52Ap81.Nf2hpA9*>
Record-Route: <sip:209.###.###.###;r2=on;lr=on;ftag=as471a1f75>
Record-Route: <sip:64.###.###.###;lr;ftag=as471a1f75>
From: "Anonymous" <sip:[email protected]:5060>;tag=as471a1f75
To: <sip:928#######@64.###.###.###:5060>;tag=as199dc3d1
Call-ID: [email protected].###.###:5060
CSeq: 102 INVITE
Server: Asterisk PBX 16.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Contact:
<sip:209.###.###.###;line=sr-1RaGXx7VX8CAKx1yp8oFKfFqKfFqKfFqKfFVXx9GpxF*>
Content-Type: application/sdp
Content-Length: 274
v=0
o=root 1644013823 1644013823 IN IP4 209.###.###.###
s=Asterisk PBX 16.18.0
c=IN IP4 209.###.###.###
t=0 0
m=audio 19180 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=nortpproxy:yes
ACK
sip:209.###.###.###;line=sr-1RaGXx7VX8CAKx1yp8oFKfFqKfFqKfFqKfFVXx9GpxF* SIP/2.0
Via: SIP/2.0/UDP 64.###.###.###:5060;branch=z9hG4bKf229.9bb425c2.2
Via: SIP/2.0/UDP
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bKvtgb6f1048h1g6l9s890.1
Max-Forwards: 67
From: "Anonymous" <sip:[email protected]:5060>;tag=as471a1f75
To: <sip:928#######@64.###.###.###:5060>;tag=as199dc3d1
Contact: <sip:anonymous@206.###.###.###:5060;transport=udp>
Call-ID: [email protected].###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route: <sip:209.###.###.###;r2=on;lr=on;ftag=as471a1f75>
Route:
<sip:209.###.###.###;line=sr-1RaGXxdGcxdGcxdGcxgTp8eVKxT-jxeE1xT-jxehH02vI52Ap81.Nf2hpA9*>
--
^C
On 1/16/22 3:16 PM, Ovidiu Sas wrote:
> Use your 209.x external IP.
>
> On Sun, Jan 16, 2022 at 18:07 Chad <[email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>> wrote:
>
> Yes I am using a 172.16.x.x IP and it works, it rewrites the
headers, but again because 172.16.x.x is also a
private IP
> it is the same as using my real 10.x.x.x IP. The carrier's ACK
throws away the local IP and sends the
response to my
> 209.x external IP.
>
>
> --
> ^C
>
>
> On 1/16/22 1:38 PM, Ovidiu Sas wrote:
> > Have you tried using the mask_ip param:
> >
https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip>
>
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip>>
> >
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip>
>
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip
<https://www.kamailio.org/docs/modules/devel/modules/topoh.html#topoh.p.mask_ip>>>
> >
> > -ovidiu
> >
> > On Sun, Jan 16, 2022 at 16:09 Chad <[email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> >
> > I found a sample config file using topoh, which I copied
(with some changes) and added the topoh
module to my
> config.
> > It works fine, but it does not solve the problem.
> > In fact it has the exact same problem, because all the topoh
module does is replace one private IP with
> another in the
> > 2nd (top most) Record-Route header.
> > So the carrier still changes the ACK to the public IP and the
call is still broken in the exact same way.
> > It was super easy to add, but does not work, 1 possible
solution down.
> >
> > --
> > ^C
> >
> >
> > On 1/16/22 8:26 AM, Ovidiu Sas wrote:
> > > Most of the time, if you get the right person on the
carrier's side
> > > and you explain the situation, they will come up with a
solution.
> > > If not, you need to break the RFC in a way that will
counterpart their breakage.
> > >
> > > The carrier is also using a SIP proxy (maybe kamailio, who
knows).
> > > In the old days, the default kamailio config was using
> > > fix_nated_contact() to deal with NATed devices and this is
exactly the
> > > behavior that you are seeing.
> > > The recommended way to deal with NATed devices is to use
> > > add_contact_alias([ip_addr, port, proto]) which is RFC
compliant.
> > >
> > > There are several solution for this scenario:
> > > - mangle the signaling to allow proper routing on your
end
> > > - use a B2BUA in between your kamailio and carrier
> > > - configure kamailio to use one of the topology hiding
modules:
> > > topoh, topos, topos_redis
> > > - maybe something else ... :)
> > >
> > > There's no right or wrong approach, one must be
comfortable with the
> > > chosen solution to be able to maintain it.
> > >
> > > -ovidiu
> > >
> > > On Sat, Jan 15, 2022 at 9:14 PM Chad <[email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>
> > >> Ok so in short I was not doing anything wrong (although I
had some miss-configurations), but the
carrier is
> > (i.e. they
> > >> are a bad actor). When they said I was doing it wrong,
they did not mean in the RFC sense they
meant in
> the "to work
> > >> with us" sense. Now in order for me to get it to work
with their SBC I have to mangle the contact
on the
> way out an
> > >> unmangle it on the return in Kamailio somehow, as I
originally purposed.
> > >> However I have no idea how to do that :)
> > >>
> > >> Shouldn't we (the Kamailio community) assume there are
lots of bad actors out there and possibly many
> Kamailio users
> > >> with this exact same issue (I personally know of at least
2 bad actor carriers right now) and
create some
> kind of
> > >> template or snippet that we can publicly publish on the
Kamailio docs or wiki for all of the Kamailio
> community
> > to use
> > >> for this use case?
> > >>
> > >> I have been fighting with carriers about this for years
and they always said I was doing it wrong
and I don't
> > know the
> > >> SIP RFC well enough to fight back. So why not build a
solution for everyone out there that has to
deal with a
> > bad actor?
> > >>
> > >> --
> > >> ^C
> > >>
> > >>
> > >> On 1/15/22 11:40 AM, Ovidiu Sas wrote:
> > >>> As expected, your carrier is bogus and "thinks" it knows
better.
> > >>> Your carrier is treating your setup as a dumb endpoint
and is
> > >>> re-writing the Contact header:
> > >>> You provide this contact header in 200 OK:
> > >>> Contact: <sip:928#######@10.###.###.104:5060>
> > >>> The carrier should set the RURI in ACK like this:
> > >>> ACK sip:928#######@10.###.###.104:5060 SIP/2.0
> > >>> Instead, your ACK is sent to you like this:
> > >>> ACK sip:928#######@209.###.###.###:5060 SIP/2.0
> > >>>
> > >>> The RURI in ACK should point to the private IP of the
asterisk server,
> > >>> not to the public IP of the kamailio server.
> > >>> You need to ask the carrier to follow the SIP RFC and
not treat your
> > >>> endpoints like dumb SIP endpoints.
> > >>>
> > >>> There's a high chance that they won't do it :)
> > >>> Your best chance is to manually mangle the URI in
Contact in the 200
> > >>> OK in a way that when you receive the ACK with the
mangled RURI, you
> > >>> can restore the original URI and let kamailio do the
proper routing to
> > >>> the private IP of the asterisk serverr.
> > >>> You should be able to achieve this by using one of the
following functions:
> > >>>
https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact>
>
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact>>
> >
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact>
>
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/mangler.html#mangler.f.encode_contact>>>
> > >>>
https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact>
>
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact>>
> >
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact>
>
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.encode_contact>>>
> > >>>
https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode>
>
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode>>
> >
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode>
>
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode
<https://kamailio.org/docs/modules/5.5.x/modules/siputils.html#siputils.f.contact_param_encode>>>
> > >>>
> > >>> Regards,
> > >>> Ovidiu Sas
> > >>>
> > >>> On Sat, Jan 15, 2022 at 1:28 PM Chad <[email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>>>
> > >>>> I changed the listen per your advice and here is the
200 and ACK.
> > >>>> I get no audio and the the call disconnects and I see
this is the Asterisk log:
> > >>>> [Jan 15 10:17:13] WARNING[29953] chan_sip.c:
Retransmission timeout reached on transmission
> > >>>> [email protected]:5060
<http://[email protected]:5060>
> <http://[email protected]:5060
<http://[email protected]:5060>>
> > <http://[email protected]:5060
<http://[email protected]:5060>
> <http://[email protected]:5060
<http://[email protected]:5060>>> for seqno
102 (Critical Response) -- See
> > >>>>
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>
> <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>>
> >
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>
> <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>>>
> > >>>> Packet timed out after 6401ms with no response
> > >>>> [Jan 15 10:17:13] WARNING[29953] chan_sip.c: Hanging up
call
> > [email protected]:5060
<http://[email protected]:5060>
<http://[email protected]:5060
<http://[email protected]:5060>>
> <http://[email protected]:5060
<http://[email protected]:5060>
> <http://[email protected]:5060
<http://[email protected]:5060>>> - no
> > >>>> reply to our critical packet (see
https://wiki.asterisk.org/wik <https://wiki.asterisk.org/wik>
<https://wiki.asterisk.org/wik <https://wiki.asterisk.org/wik>>
> <https://wiki.asterisk.org/wik <https://wiki.asterisk.org/wik>
<https://wiki.asterisk.org/wik
<https://wiki.asterisk.org/wik>>>
> > >>>>
> > >>>> FYI 10.###.###.254 is the private virtual IP on the
Kamailio server and 10.###.###.104 is the
asterisk box.
> > >>>>
> > >>>> SIP/2.0 200 OK
> > >>>> Via: SIP/2.0/UDP
64.###.###.###:5060;branch=z9hG4bK26ab.5547ac15.0
> > >>>> Via: SIP/2.0/UDP
>
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bK6gj48a00dolcl3jm2gq0.1
> > >>>> Record-Route:
<sip:10.###.###.254;r2=on;lr=on;ftag=as04035ef0>
> > >>>> Record-Route:
<sip:209.###.###.###;r2=on;lr=on;ftag=as04035ef0>
> > >>>> Record-Route: <sip:64.###.###.###;lr;ftag=as04035ef0>
> > >>>> From: "Anonymous"
<sip:[email protected]:5060>;tag=as04035ef0
> > >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> > >>>> Call-ID:
[email protected].###.###:5060
> > >>>> CSeq: 102 INVITE
> > >>>> Server: Asterisk PBX 16.18.0
> > >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> > >>>> Supported: replaces, timer
> > >>>> Contact: <sip:928#######@10.###.###.104:5060>
> > >>>> Content-Type: application/sdp
> > >>>> Content-Length: 274
> > >>>>
> > >>>> v=0
> > >>>> o=root 1911037741 1911037741 IN IP4 209.###.###.###
> > >>>> s=Asterisk PBX 16.18.0
> > >>>> c=IN IP4 209.###.###.###
> > >>>> t=0 0
> > >>>> m=audio 11384 RTP/AVP 0 101
> > >>>> a=rtpmap:0 PCMU/8000
> > >>>> a=rtpmap:101 telephone-event/8000
> > >>>> a=fmtp:101 0-16
> > >>>> a=ptime:20
> > >>>> a=maxptime:150
> > >>>> a=sendrecv
> > >>>> a=nortpproxy:yes
> > >>>>
> > >>>> ACK sip:928#######@209.###.###.###:5060 SIP/2.0
> > >>>> Via: SIP/2.0/UDP
64.###.###.###:5060;branch=z9hG4bK26ab.5547ac15.2
> > >>>> Via: SIP/2.0/UDP
>
206.###.###.###:5060;rport=5060;received=206.###.###.###;branch=z9hG4bK91l3it006gr9oiulcqn0.1
> > >>>> Max-Forwards: 67
> > >>>> From: "Anonymous"
<sip:[email protected]:5060>;tag=as04035ef0
> > >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> > >>>> Contact:
<sip:anonymous@206.###.###.###:5060;transport=udp>
> > >>>> Call-ID:
[email protected].###.###:5060
> > >>>> CSeq: 102 ACK
> > >>>> User-Agent: packetrino
> > >>>> Content-Length: 0
> > >>>> Route: <sip:209.###.###.###;r2=on;lr=on;ftag=as04035ef0>
> > >>>> Route: <sip:10.###.###.254;r2=on;lr=on;ftag=as04035ef0>
> > >>>>
> > >>>>
> > >>>> --
> > >>>> ^C
> > >>>>
> > >>>>
> > >>>> On 1/15/22 10:21 AM, Ovidiu Sas wrote:
> > >>>>> This is false. The IP in the Contact header must be
routable by the
> > >>>>> SIP hop from the top Record-Route header in the reply.
> > >>>>> The carrier (and it seems that they have a PROXY also)
must be able to
> > >>>>> route to their adjacent SIP hop, which is your public
IP (the IP in
> > >>>>> the second Record-Route header).
> > >>>>> It seems that the carrier is not taking into account
that they might
> > >>>>> interface with other proxies.
> > >>>>> Most likely, your carrier expects to interface with a
simple SIP UA,
> > >>>>> not with another proxy. This is a pretty common setup
for most of the
> > >>>>> carriers, although many new carrier implementations
are taking care of
> > >>>>> the proxy to proxy calls.
> > >>>>>
> > >>>>> It would be helpful to see the ACK that is sent by the
carrier in
> > >>>>> response to your 200ok (after you fix your config and
you have your
> > >>>>> private IP listed in the Record-Route header).
> > >>>>>
> > >>>>> -ovidiu
> > >>>>>
> > >>>>> On Sat, Jan 15, 2022 at 12:33 PM Chad <[email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>>>>>
> > >>>>>> Hmm, I don't think you are right that the Contact
header can be a private IP even if the RR is
correct.
> > >>>>>> I did some research on it and I found several places
saying it must be a routable IP which is
what the
> > carrier also said.
> > >>>>>>
> > >>>>>> "The Contact header contains the SIP URI where the
client wants to be contacted for subsequent
requests.
> > That means that
> > >>>>>> the host part of the URI must be globally reachable
by anyone.
> > >>>>>> If your contact contains a private IP (behind a NAT?)
then it is wrong, because other peers cannot
> reach you
> > with that."
> > >>>>>>
> > >>>>>>
> > >>>>>> --
> > >>>>>> ^C
> > >>>>>>
> > >>>>>>
> > >>>>>> On 1/15/22 9:05 AM, Ovidiu Sas wrote:
> > >>>>>>> You have a different problem then.
> > >>>>>>> Having private IPs in Contact is fine. You need to
lose route the
> > >>>>>>> calls (kamailio will add two Record-Route headers)
and the origination
> > >>>>>>> server will set the RURI to the private IP from
Contact, but it will
> > >>>>>>> send the in-dialog requests to the public IP of
kamailio. This has
> > >>>>>>> nothing to do with virtual IPs.
> > >>>>>>> Maybe you have a buggy client that doesn't do proper
loose routing.
> > >>>>>>>
> > >>>>>>> -ovidiu
> > >>>>>>>
> > >>>>>>> On Sat, Jan 15, 2022 at 11:50 AM Chad
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>>>>>>>
> > >>>>>>>> Ovidiu,
> > >>>>>>>> Thank you again for your response.
> > >>>>>>>> One is public (an internet IP) and one is private
(a 10.x ip).
> > >>>>>>>> Apparently this is a known problem with virtual
IPs, it does not work.
> > >>>>>>>> When the asterisk server responds to the invite it
sends a contact header with the private
IP and
> Kamailio
> > does not
> > >>>>>>>> rewrite it to the advertised public IP. So the
originating server sees the private IP in the
Contact
> > header and tries to
> > >>>>>>>> send the traffic to the 10.x IP (which is
non-routable) and the call dies.
> > >>>>>>>> I have been trying things for a long time to fix
this (years) what you are saying will not
fix it
> because
> > of the virtual
> > >>>>>>>> IPs.
> > >>>>>>>> If it was a normal IP it would work fine. It has
something to do with the routing table and
how mhomed
> > detects networks.
> > >>>>>>>>
> > >>>>>>>> --
> > >>>>>>>> ^C
> > >>>>>>>>
> > >>>>>>>>
> > >>>>>>>> On 1/15/22 8:36 AM, Ovidiu Sas wrote:
> > >>>>>>>>> Hello Chad,
> > >>>>>>>>>
> > >>>>>>>>> The floating IPs that you have, are they both
private IPs or one
> > >>>>>>>>> private IP and the other one a public IP?
> > >>>>>>>>>
> > >>>>>>>>> If you have to two floating private IPs, then you
need a config like this:
> > >>>>>>>>> listen=FLOATING_UDP_PRIVATE1 advertise
PUBLIC_UDP_IP
> > >>>>>>>>> listen=FLOATING_UDP_PRIVATE2
> > >>>>>>>>>
> > >>>>>>>>> In the config, before relaying the initial INVITE
you need to detect
> > >>>>>>>>> the direction of the call and set $fs accordingly:
> > >>>>>>>>> if (CAL_FROM_PRIVATE_TO_PUBLIC) {
> > >>>>>>>>> $fs = udp:FLOATING_UDP_PRIVATE1
> > >>>>>>>>> }
> > >>>>>>>>> else {
> > >>>>>>>>> $fs = udp:FLOATING_UDP_PRIVATE2
> > >>>>>>>>> }
> > >>>>>>>>>
> > >>>>>>>>> If you have a floating private IPs and a floating
public IP, then you
> > >>>>>>>>> need a config like this:
> > >>>>>>>>> listen=FLOATING_UDP_PRIVATE
> > >>>>>>>>> listen=FLOATING_UDP_PUBLIC
> > >>>>>>>>>
> > >>>>>>>>> There should be no need to force the socket, but
if you do, there's no
> > >>>>>>>>> harm (actually it's better and faster).
> > >>>>>>>>>
> > >>>>>>>>> Hope this clarifies things and helps,
> > >>>>>>>>> -ovidiu
> > >>>>>>>>>
> > >>>>>>>>> On Sat, Jan 15, 2022 at 9:48 AM Chad
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>>>>>>>>>
> > >>>>>>>>>> Ovidiu,
> > >>>>>>>>>> Thank you for your response.
> > >>>>>>>>>>
> > >>>>>>>>>> I have done that, in addition to the linux
ip_nonlocal_bind I have also set the Kamailio
> ip_free_bind=1
> > and it does not
> > >>>>>>>>>> work.
> > >>>>>>>>>> Here are my relevant config lines:
> > >>>>>>>>>> listen=LISTEN_UDP_PRIVATE advertise
MY_PUBLIC_IP:5060
> > >>>>>>>>>> listen=LISTEN_UDP_PUBLIC
> > >>>>>>>>>>
> > >>>>>>>>>> mhomed=1
> > >>>>>>>>>> ip_free_bind=1
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >>>>>>>>>> In my /etc/sysctl.conf I have (yes I applied it
with sysctl -p, and I have been using it for a
> long time
> > and have
> > >>>>>>>>>> rebooted as well):
> > >>>>>>>>>> net.ipv4.ip_nonlocal_bind=1
> > >>>>>>>>>> --
> > >>>>>>>>>> ^C
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >>>>>>>>>> On 1/15/22 4:55 AM, Ovidiu Sas wrote:
> > >>>>>>>>>>> Hello Chad,
> > >>>>>>>>>>>
> > >>>>>>>>>>> You can add a listen directive to your config
for the virtual IPs
> > >>>>>>>>>>> (both public and private) and then you don't
need to manually modify
> > >>>>>>>>>>> any headers or use force_send_socket().
> > >>>>>>>>>>> You need to enable non local IP binding so
kamailio can start on the
> > >>>>>>>>>>> server that doesn't have the virtual IP:
> > >>>>>>>>>>> echo 1 > /proc/sys/net/ipv4/ip_nonlocal_bind
> > >>>>>>>>>>> To make the change permanent, edit your
sysctl.conf file and enable it there:
> > >>>>>>>>>>> net/ipv4/ip_nonlocal_bind = 1
> > >>>>>>>>>>>
> > >>>>>>>>>>> Regards
> > >>>>>>>>>>> Ovidiu Sas
> > >>>>>>>>>>>
> > >>>>>>>>>>>
> > >>>>>>>>>>> On Sat, Jan 15, 2022 at 4:16 AM Chad
<[email protected] <mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected] <mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>>> wrote:
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> We are looking for some help (possibly a paid
consultant) to help us with our Kamailio
setup.
> > >>>>>>>>>>>> To keep this as short as possible: we use
Kamailio as a NAT proxy to bridge our external
IP and our
> > private IP asterisk
> > >>>>>>>>>>>> servers (via dispatcher).
> > >>>>>>>>>>>> However both the external IP and the internal
IP that the Kamailio server uses are
virtual IPs
> created
> > by keepalived.
> > >>>>>>>>>>>> Because of that neither mhomed nor
fix_nated_contact work, and we use force_send_socket to
> direct the
> > traffic.
> > >>>>>>>>>>>> We run linux Debian 10 for the OS.
> > >>>>>>>>>>>> Also we do not use a DB at all, everything is
done with local config files.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> The problem is that when traffic goes out the
Contact header has a private IP in it, like:
> > >>>>>>>>>>>> Contact: <sip:##########@10.10.10.###]:5060
<http://10.10.10.#%23%23]:5060>
<http://10.10.10.#%23%23]:5060 <http://10.10.10.#%23%23]:5060>>
> <http://10.10.10.#%23%23]:5060 <http://10.10.10.#%23%23]:5060>
<http://10.10.10.#%23%23]:5060
<http://10.10.10.#%23%23]:5060>>>>
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> There are 2 possible solutions to this:
> > >>>>>>>>>>>> 1. Make changes to linux, keepalived and/or
Kamailio so that Kamailio recognize the
virtual IPs so
> > that mhomed and
> > >>>>>>>>>>>> fix_nated_contact work as usual.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> 2. Create a manual header rewrite system.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> If solution #2:
> > >>>>>>>>>>>> What we need to do is create a way to rewrite
the contact header to the external IP on
the way out,
> > and on the way back
> > >>>>>>>>>>>> rewrite it back to the internal server that the
call is already connected to.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> Not sure if we will need to store those paths
on the server or if we can do some kind of
cheat with
> > another persistant
> > >>>>>>>>>>>> header like P-Preferred-Identity or
P-Asserted-Identity (i.e. store the internal IP in
the name
> field
> > or something).
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> If anyone out there know of a way to do this or
wants to give it a try please reach out
to me.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> Thank you all for your time.
> > >>>>>>>>>>>>
> > >>>>>>>>>>>> --
> > >>>>>>>>>>>> ^C
> > >>>>>>>>>>>> Chad
> > >>>>>>>>>>>>
> > >>>>>>>>>>>>
__________________________________________________________
> > >>>>>>>>>>>> Kamailio - Users Mailing List - Non Commercial
Discussions
> > >>>>>>>>>>>> * [email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected]
<mailto:[email protected]> <mailto:[email protected]
<mailto:[email protected]>>>
> > >>>>>>>>>>>> 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
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>>
> > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>>>
> > >>>>>>>>>>>
> > >>>>>>>>>>>
> > >>>>>>>>>>>
> > >>>>>>>>>>> --
> > >>>>>>>>>>> VoIP Embedded, Inc.
> > >>>>>>>>>>> http://www.voipembedded.com
<http://www.voipembedded.com> <http://www.voipembedded.com
<http://www.voipembedded.com>> <http://www.voipembedded.com
<http://www.voipembedded.com>
> <http://www.voipembedded.com <http://www.voipembedded.com>>>
> > >>>>>>>>>>>
> > >>>>>>>>>>>
__________________________________________________________
> > >>>>>>>>>>> Kamailio - Users Mailing List - Non Commercial
Discussions
> > >>>>>>>>>>> * [email protected]
<mailto:[email protected]>
<mailto:[email protected] <mailto:[email protected]>>
> <mailto:[email protected]
<mailto:[email protected]> <mailto:[email protected]
<mailto:[email protected]>>>
> > >>>>>>>>>>> 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
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>>
> > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
> <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>>>
> > >>>>>>>>>
> > >>>>>>>>>
> > >>>>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>
> > >>>
> > >>>
> > >
> > >
> > >
> >
> > --
> > VoIP Embedded, Inc.
> > http://www.voipembedded.com <http://www.voipembedded.com>
<http://www.voipembedded.com
<http://www.voipembedded.com>> <http://www.voipembedded.com
<http://www.voipembedded.com>
<http://www.voipembedded.com <http://www.voipembedded.com>>>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com <http://www.voipembedded.com>
<http://www.voipembedded.com <http://www.voipembedded.com>>
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* [email protected] <mailto:[email protected]>
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
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Regards,
David Villasmil
email: [email protected] <mailto:[email protected]>
phone: +34669448337
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* [email protected]
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
