Re: [SR-Users] kamailio log dialog correlation
Are you running 5.5?
Do you have log_prefix_mode=1
If set to 1, then the log prefix is evaluated before/after each config action (needs to be set when the log_prefix has
variables that are different based on the context of config execution, e.g., $cfg(line)).
BTW "grep cfgutils /etc/kamailio/kamailio.cfg" may not be good enough, because the loadmodule "cfgutils.so" might be
inside of an if statement.
You also need to make sure it is actually loaded above the log_prefix line
where you use it.
I don't know how to show the loaded modules, probably some kamcli command will show you, did you search the internet for
the answer to that?
--
^C
On 1/24/22 7:41 AM, marek wrote:
[root@sbc~]# grep cfgutils /etc/kamailio/kamailio.cfg
loadmodule "cfgutils.so"
is there a way how can i check if module is loaded without errors? (something like
Asterisk's "module show like")
kamailio 5.5.3
Marek
Dne 22/01/2022 v 17:27 Chad napsal(a):
loadmodule "cfgutils.so"
--
^C
On 1/22/22 2:36 AM, marek wrote:
thanks
now i understand difference between log_prefix and modparam("xlog", "prefix",
"something")
so i'm trying this
log_prefix="{D$dlg(h_id) $cfg(route)}"
but i got this
ERROR: [core/pvapi.c:924]: pv_parse_spec2(): error searching pvar "cfg"
ERROR: [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [r/114] in
[$cfg(route)] at [5 (5)]
ERROR: [core/dprint.c:463]: log_prefix_init(): wrong format[{D$dlg(h_id)
$cfg(route)}]
it's the same if its used in
xlog("L_INFO", "D$dlg(h_id) $cfg(route) some log")
do you think there is a module missing?
Marek
Dne 19/01/2022 v 16:23 Henning Westerholt napsal(a):
Hello,
you could try to use log_prefix. Have not tried it yet with a dialog variable.
http://www.kamailio.org/wiki/cookbooks/5.5.x/core#log_prefix
Regarding output of a line number, you could try to use this one:
http://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#cfg_key_-_config_file_attributes
Cheers,
Henning
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Re: [SR-Users] kamailio log dialog correlation
loadmodule "cfgutils.so"
--
^C
On 1/22/22 2:36 AM, marek wrote:
thanks
now i understand difference between log_prefix and modparam("xlog", "prefix",
"something")
so i'm trying this
log_prefix="{D$dlg(h_id) $cfg(route)}"
but i got this
ERROR: [core/pvapi.c:924]: pv_parse_spec2(): error searching pvar "cfg"
ERROR: [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [r/114] in
[$cfg(route)] at [5 (5)]
ERROR: [core/dprint.c:463]: log_prefix_init(): wrong format[{D$dlg(h_id)
$cfg(route)}]
it's the same if its used in
xlog("L_INFO", "D$dlg(h_id) $cfg(route) some log")
do you think there is a module missing?
Marek
Dne 19/01/2022 v 16:23 Henning Westerholt napsal(a):
Hello,
you could try to use log_prefix. Have not tried it yet with a dialog variable.
http://www.kamailio.org/wiki/cookbooks/5.5.x/core#log_prefix
Regarding output of a line number, you could try to use this one:
http://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#cfg_key_-_config_file_attributes
Cheers,
Henning
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Re: [SR-Users] kamailio log dialog correlation
I wanted the same thing only with line number, in the end I had to do a global search and replace, I could not get it to
work via a global param.
I even tried xlogl which just threw errors that no such command existed (even though it is listed in the docs:
https://kamailio.org/docs/modules/5.0.x/modules/xlog.html#xlog.f.xlogl).
Here is what I ended up with:
xlog("L_INFO", "### Something to log ### Line: $cfg(line)\n");
Maybe someone else knows more, but I could not get it to work without
hardcoding it into every xlog call.
--
^C
On 1/19/22 7:06 AM, marek wrote:
hi,
is it possible create log of SIP dialog like
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ?
it looks like its possible like this
xlog("L_INFO", "D$dlg(h_id) something to log \n");
but i must copy $dlg(h_id) in every xlog
tried this
modparam("xlog", "prefix", "D$dlg(h_id)")
but it not works with variables
any other tips/ideas?
goal is to correlate logs for one dialog (call)
thanks
Marek
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Re: [SR-Users] Help with rewriting headers for NAT manually
David, Thank you for the suggestion. Do you have any sample config files you can point me at? -- ^C On 1/17/22 12:41 AM, David Villasmil wrote: Take a look at freeSWITCH On Mon, 17 Jan 2022 at 00:58, Chad mailto:ccolu...@hotmail.com>> 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: Record-Route: Record-Route: From: "Anonymous" ;tag=as471a1f75 To: ;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: 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" ;tag=as471a1f75 To: ;tag=as199dc3d1 Contact: Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@10.44.###.###:5060 CSeq: 102 ACK User-Agent: packetrino Content-Length: 0 Route: Route: -- ^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 mailto:ccolu...@hotmail.com> <mailto:ccolu...@hotmail.com <mailto:ccolu...@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>> > > <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 mailto:ccolu...@hotmail.com> <mailto:ccolu...@hotmail.com <mailto:ccolu...@hotmail.com>> > <mailto:ccolu...@hotmail.com <mailto:ccolu...@hotmail.com> <mailto:ccolu...@hotmail.com <mailto:ccolu...@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 t
Re: [SR-Users] Help with rewriting headers for NAT manually
Michael, Thank you for the feedback. -- ^C On 1/16/22 4:02 PM, Michael Young wrote: Chad, In my experience, if you carrier partner is one of the bigger carriers, and\or you use multiple carriers, Kamailio is the best solution available. With some of the smaller carriers\resellers it just makes more sense to use Asterisk rather than argue with them about how they are ignoring and breaking RFCs. While Asterisk can be inefficient, it generally "just works" in those situations. Based on what I have read of your situation in the list I think I can guess which company you are working with. Their "SBC" is Freeswitch-based. I have had a similar debate with them about RFCs, and yes, you are better off with Asterisk in that case. Michael On 1/16/2022 5:20 PM, Chad wrote: I have been reading a lot more about the problem and it seems my mangle/unmangle solution is basically B2BUA. So I need a B2BUA solution and it seems like Kamailio does not really do B2BUA. Instead of installing something else I don't know (SEMS or Sippy), it makes more sense to find something that can handle it all. I have read that opensips has B2BUA functionality built in, so I am seriously considering simply replacing Kamailio with opensips. In reality my system has such a low load I can probably replace Kamailio with Asterisk as a B2BUA and it would be fine, but from what I have read Asterisk is very inefficient for B2BUA. -- ^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> -ovidiu On Sun, Jan 16, 2022 at 16:09 Chad mailto:ccolu...@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 mailto:ccolu...@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? >>
Re: [SR-Users] Help with rewriting headers for NAT manually
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:
Record-Route:
Record-Route:
From: "Anonymous" ;tag=as471a1f75
To: ;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:
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" ;tag=as471a1f75
To: ;tag=as199dc3d1
Contact:
Call-ID: 3510f7167e1a0f6a5423234b1d176a8b@10.44.###.###:5060
CSeq: 102 ACK
User-Agent: packetrino
Content-Length: 0
Route:
Route:
--
^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 mailto:ccolu...@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 mailto:ccolu...@hotmail.com>
<mailto:ccolu...@hotmail.com <mailto:ccolu...@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:
Re: [SR-Users] Help with rewriting headers for NAT manually
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 mailto:ccolu...@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 mailto:ccolu...@hotmail.com> <mailto:ccolu...@hotmail.com <mailto:ccolu...@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 mailto:ccolu...@hotmail.com> <mailto:ccolu...@hotmail.com <mailto:ccolu...@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 >
Re: [SR-Users] Help with rewriting headers for NAT manually
I have been reading a lot more about the problem and it seems my mangle/unmangle solution is basically B2BUA. So I need a B2BUA solution and it seems like Kamailio does not really do B2BUA. Instead of installing something else I don't know (SEMS or Sippy), it makes more sense to find something that can handle it all. I have read that opensips has B2BUA functionality built in, so I am seriously considering simply replacing Kamailio with opensips. In reality my system has such a low load I can probably replace Kamailio with Asterisk as a B2BUA and it would be fine, but from what I have read Asterisk is very inefficient for B2BUA. -- ^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> -ovidiu On Sun, Jan 16, 2022 at 16:09 Chad mailto:ccolu...@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 mailto:ccolu...@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: >>> 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
Re: [SR-Users] Help with rewriting headers for NAT manually
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> -ovidiu On Sun, Jan 16, 2022 at 16:09 Chad mailto:ccolu...@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 mailto:ccolu...@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: >>> 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 th
Re: [SR-Users] Help with rewriting headers for NAT manually
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 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: 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/siputils.html#siputils.f.encode_contact 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 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 for seqno 102 (Critical Response) -- See 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 - no reply to our critical packet (see 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-
Re: [SR-Users] Help with rewriting headers for NAT manually
Shah, Thank you for your time and effort. Of course you and Ovidiu are right in that the carrier should be RFC compliant, but getting away from the technical and looking at it more from a practical point of view: 1. Fighting the carrier is problematic, they think they are right. 2. Even if they agree right now it will take time for them to make the change, test it, and release it (could be weeks or months), because they don't want to break their production system for other clients. 3. There are other bad actors out there including upstream providers (i.e. a reinvite) and other carriers we all use. Which means fighting the fight over and over with every carrier you deal with even some that you are not a direct client of (in the case of upstream) So if we (the Kamailio/OpenSIPs community) have a solution that we can implement on our end that solves these problems in all cases and does not hurt any RFC compliant carriers (except for a small resource hit) we overcome all the above problems from our side. I look at is as: if we rewrite (mangle) the Contact header to the external routable IP in all cases and reverse the Contact rewrite (unmangle) the return traffic the proxy becomes invisible to the carriers and we have 1 less thing to worry about breaking. BTW this idea comes from linux firewalls/load balancers, they have a similar solution for iptables/nftables/LVS and forwarding http traffic to a private cluster. It makes the server stack's internals invisible to the end user's browser. All they see is the single external IP and all traffic is 1-to-1 from the end user's perspective. I am happy to pitch in some financial incentive to getting a solution to my immediate problem, but I want the solution to be open source for the whole community. Thanks again! -- ^C Chad On 1/16/22 1:00 AM, Shah Hussain Khattak wrote: Hi There, If I look at the latest SIP trace you shared: 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: Record-Route: Record-Route: From: "Anonymous" ;tag=as04035ef0 To: ;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: 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" ;tag=as04035ef0 To: ;tag=as7047ed05 Contact: Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060 CSeq: 102 ACK User-Agent: packetrino Content-Length: 0 Route: Route: The ACK is getting sent to the Kamailio with correct Route information: Route: Route: The Kamailio server should strip the 1^st and 2^nd Route(s) header from the ACK and should relay it towards the next-hop as per the request URI. Please note that Kamailio is sending Double RR headers ( When a Proxy receives a request on one network interface and sends it onwards using a different interface e.g. WAN to LAN, this will normally require the addition of an extra Record-Route header. i.e. the Proxy must add two RR headers where you might normally expect it to add one.) Record-Route: Record-Route: The problem is, the peer behavior is not compliant with the specs. It is sending the ACK with RURI set to: ACK sip:928###@209.###.###.###:5060SIP/2.0 Ideally, it should have sent the ACK with the following Request-URI: ACK sip:928###@10.###.###.104:5060SIP/2.0 Once this ACK will be received on Kamailio, it will relay it towards the Asterisk IP, which is 10.###.###.104. For further understanding of the ACK routing, you can refer to the following post: https://lists.cs.columbia.edu/pipermail/sip-implementors/2019-February/031229.html <https://lists.cs.columbia.edu/pipermail/sip-implementors/2019-February/031229.html> The peer is not copying the 200 OK Contact header URI into the ACK message and it is a problem. Lastly, the trace might be showing only part of the puzzle, it is also suggested to get a capture on the remote peer end, because it is sending the 209.###.###.### IP in the ACK, which seems to be the public interface of the Kamailio server. I am not sure if there is some device in the path, that is changing the contact IP in the 200 OK to the Kamailio public IP? Ovidiu also explained th
Re: [SR-Users] Help with rewriting headers for NAT manually
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: 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/siputils.html#siputils.f.encode_contact 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 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 for seqno 102 (Critical Response) -- See 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 - no reply to our critical packet (see 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: Record-Route: Record-Route: From: "Anonymous" ;tag=as04035ef0 To: ;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: 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" ;tag=as04035ef0 To: ;tag=as7047ed05 Contact: Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060 CSeq: 102 ACK User-Agent: packetrino Content-Length: 0 Route: Route: -- ^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 carrie
Re: [SR-Users] Help with rewriting headers for NAT manually
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 for seqno 102 (Critical Response) -- See 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 - no reply to our critical packet (see 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: Record-Route: Record-Route: From: "Anonymous" ;tag=as04035ef0 To: ;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: 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" ;tag=as04035ef0 To: ;tag=as7047ed05 Contact: Call-ID: 5ab1525b3712f34c2ab272ae55e649e5@10.44.###.###:5060 CSeq: 102 ACK User-Agent: packetrino Content-Length: 0 Route: Route: -- ^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 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 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 n
Re: [SR-Users] Help with rewriting headers for NAT manually
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 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 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 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:
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
Re: [SR-Users] Help with rewriting headers for NAT manually
It would be great if you are right and I am simply doing something else wrong
in the config file!
Here is the 200 OK (note I have enable_double_rr enabled):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 18.###.###.###:5060;branch=z9hG4bK22b2.6b6d30e5.0
Via: SIP/2.0/UDP 66.###.###.###:5060;branch=z9hG4bK22b2.d15ac8a.0
Record-Route:
Record-Route:
Record-Route:
From: ;tag=gK0e16642e
To: ;tag=as488a6fb4
Call-ID: 202251204_54250714@206.###.###.###
CSeq: 710596 INVITE
Server: Asterisk PBX 16.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Require: timer
Content-Length: 272
v=0
o=root 153822920 153822920 IN IP4 209.###.###.###
s=Asterisk PBX 16.18.0
c=IN IP4 209.###.###.###
t=0 0
m=audio 17198 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
--
^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 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 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 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:
There are 2 possible solutions to this:
1. Make changes to linux, keepalived and/or Kamailio so that Kamai
Re: [SR-Users] Help with rewriting headers for NAT manually
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 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 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:
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
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Re: [SR-Users] Help with rewriting headers for NAT manually
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 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: 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@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 -- VoIP Embedded, Inc. http://www.voipembedded.com __ 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 __ 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
[SR-Users] Help with rewriting headers for NAT manually
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: 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@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
[SR-Users] Connect SIP Trunk to PBX
Hello, I hope you can help, We are trying to connect our SIP trunk to our new PBX system, however our SIP trunk provider requires a static IP to add to the endpoint. Our new PBX provider uses a range of IPs that are connected to a domain and the IPs change all the time. Is there a way of using Kamailio that we could get our Sip Trunk provider and PBX system to work together? Regards ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
