[SR-Users] Help with rewriting headers for NAT manually
Chad
ccolumbu at hotmail.com
Mon Jan 17 00:21:33 CET 2022
If I use my external IP do I turn off enable_double_rr?
--
^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 <ccolumbu at hotmail.com <mailto:ccolumbu at 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_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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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.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.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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 at 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 at 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 at anonymous.invalid:5060>;tag=as04035ef0
> > >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> > >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5 at 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 at anonymous.invalid:5060>;tag=as04035ef0
> > >>>> To: <sip:928#######@64.###.###.###:5060>;tag=as7047ed05
> > >>>> Contact: <sip:anonymous at 206.###.###.###:5060;transport=udp>
> > >>>> Call-ID: 5ab1525b3712f34c2ab272ae55e649e5 at 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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 <ccolumbu at hotmail.com <mailto:ccolumbu at hotmail.com>
> <mailto:ccolumbu at hotmail.com <mailto:ccolumbu at 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 at lists.kamailio.org <mailto:sr-users at lists.kamailio.org>
> <mailto:sr-users at lists.kamailio.org <mailto:sr-users at 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 at lists.kamailio.org <mailto:sr-users at lists.kamailio.org>
> <mailto:sr-users at lists.kamailio.org <mailto:sr-users at 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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