Re: [SR-Users] Help with rewriting headers for NAT manually

16 Jan 2022

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:anonymous@anonymous.invalid:5060>;tag=as471a1f75
To: <sip:928#######@64.###.###.###:5060>;tag=as199dc3d1
Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@10.44.###.###: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:anonymous@anonymous.invalid:5060>;tag=as471a1f75
To: <sip:928#######@64.###.###.###:5060>;tag=as199dc3d1
Contact: <sip:anonymous@206.###.###.###:5060;transport=udp>
Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@10.44.###.###: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 &lt;ccolumbu(a)hotmail.com
<mailto:ccolumbu@hotmail.com>> 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…     
<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 &lt;ccolumbu(a)hotmail.com
<mailto:ccolumbu@hotmail.com>      <mailto:ccolumbu@hotmail.com
<mailto:ccolumbu@hotmail.com>>> 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 &lt;ccolumbu(a)hotmail.com
<mailto:ccolumbu@hotmail.com>      <mailto:ccolumbu@hotmail.com
<mailto:ccolumbu@hotmail.com>>> 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.enco…

<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.en…

<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.co…

<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 &lt;ccolumbu(a)hotmail.com
<mailto:ccolumbu@hotmail.com>      <mailto:ccolumbu@hotmail.com
<mailto:ccolumbu@hotmail.com>>> 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
      >>>> 5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060     
<http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060>

 <http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060     
<http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060>> 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>>

>>> Packet timed out after 6401ms with no response
>>> [Jan 15 10:17:13] WARNING[29953] chan_sip.c: Hanging up call 
5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060
<http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060>     
<http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060
     <http://5ab1525b3712f34c2ab272ae55e649e5@10.44.109.143:5060>> - 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>>
       >>>>
      >>>> 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:anonymous@anonymous.invalid:5060>;tag=as04035ef0
      >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
      >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###: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:anonymous@anonymous.invalid:5060>;tag=as04035ef0
      >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
      >>>> Contact: <sip:anonymous@206.###.###.###:5060;transport=udp>
      >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###: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 &lt;ccolumbu(a)hotmail.com
<mailto:ccolumbu@hotmail.com>      <mailto:ccolumbu@hotmail.com
<mailto:ccolumbu@hotmail.com>>> 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
&lt;ccolumbu(a)hotmail.com <mailto:ccolumbu@hotmail.com>     
<mailto:ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>>> 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
&lt;ccolumbu(a)hotmail.com <mailto:ccolumbu@hotmail.com>     
<mailto:ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>>> 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
&lt;ccolumbu(a)hotmail.com <mailto:ccolumbu@hotmail.com>     
<mailto:ccolumbu@hotmail.com <mailto:ccolumbu@hotmail.com>>> 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>>>

>>>>>>>>>>>
>>>>>>>>>>> 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
      >>>>>>>>>>>>         *
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>     
<mailto:sr-users@lists.kamailio.org <mailto: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     
<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>>

>>>>>>>>>>>
      >>>>>>>>>>>
__________________________________________________________
      >>>>>>>>>>> Kamailio - Users Mailing List - Non
Commercial Discussions
      >>>>>>>>>>>         * sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>     
<mailto:sr-users@lists.kamailio.org <mailto: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     
<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>>  
 -- 
 VoIP Embedded, Inc.
 http://www.voipembedded.com <http://www.voipembedded.com> 

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Re: [SR-Users] Help with rewriting headers for NAT manually