Hello list.
Is there a way to simplify the notation of this configuration line?
if ( (uri=~"^sip:111.*@.*") | (uri=~"^sip:222.*@.*") |
(uri=~"^sip:333.*@.*") | (uri=~"^sip:444.*@.*") | (uri=~"^sip:00.*@.*")){
with something like this maybe?
if ( (uri=~"^sip:111 | 222 | 333 | 444 | 00.*@.*")) {
Thanks.
Ricardo.-
Hi Everybody,
I'm trying to make SER work with ASTERISK, ser as a proxy server and SIP terminator, and asterisk for other services like voicemail and International calls like an UAC (thing that I know SER is not able to do). I've configured SER with the following forward route, to forward all "1" beggining call phones to forward to my asterisk for international calls:
if (uri=~"^sip:8[0-9]{7}.*") {
forward( 10.0.18.3, 5061 );
break;
}
When I restart SER, it gives me this errors:
Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: ERROR: send_rtpp_command: can't connect to RTP proxy
Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: can't get version of the RTP proxy
Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily
I've also tried with the following rewritehost but it gives me the same error:
if (uri=~"^sip:8[0-9]{7}.*") {
rewritehost("10.0.18.3, 5061");
break;
}
I know that i'm missing something with RTP that i'm still working on.
I've tried to look for it at google, but find nothing interesting.
Do anyone know what is going on ? My ser.cfg is attached.
--
Felipe Martins
Linux System Administrator
Tep Solution Provider
Mundivox Communications
Rua Lauro Muller, 116/Sala 505
RJ - Brasil - 22290-906
Tel.: 55 21 3820-8839
Fax.: 55 21 3820-8844
Hello list,
I have a question regarding to the NAT keep alive message sent by a Lynksys
PAP2 device. I configured the NAT keep Alive message as NOTIFY. When this
messsage arrives to SER the next configuration line handle it.
if (method=="NOTIFY") {
break;
};
i'm not sure if this behavior is ok. Does SER must answer to the NOTIFY
(keep alive messages).?? if the answer is yes, which message is the answer
to a NOTIFY? is better to do the NAT keep alive message with a REGISTER?
I hope that someone could help me
Ricardo Martinez.-
Hi,
I am using Windows Messenger 5.1. Do I have to use mysql database for
presence to work, or should the clients be able to see each other even
without it? Right now, I can sign in, but cannot see my contacts even though
they are online.
Thanks,
Meena.
Hi all,
If I need to shut off the Media Attribute: rtpmap:3 GSM/8000 in
SER, which file do I need to make changes in?
Also does anyone have any idea what rtpmap:101 telephone-event/8000 is?
Thanks,
Hitesh.
> Thank you first for your answer, but where can I find the error_reporting
> variable?
oh, sorry
this variable is in php.ini
> Second, in spite of changing the access permission of the ser_fifo for
> writing and reading php scripts, php scripts could not open ser_fifo for
> writing and thus got on the web browser (user_interface/my_account.php...)
> the following error "sorry -- cannot open write fifo".
> How can I solve this problem?
> Best regards,
This is a frequently asked question. The solution is in ser.cfg -
variable mod_fifo - or something like this I don't know exact name now.
Or search list archive, there are a lot of questions and answers to this.
Karel
>
> -----Ursprüngliche Nachricht-----
> Von: Karel Kozlik [mailto:kozlik@kufr.cz]
> Gesendet: Mittwoch, 22. Dezember 2004 13:07
> An: Adel Al-Hezmi
> Cc: serusers(a)lists.iptel.org
> Betreff: Re: [Serusers] Propem with the SER Web interface
>
> Hello,
> this is a problem of used library phplib (not errors, only notices).
> Will be fixed in next version of serweb. For now you can disable this
> messages by set:
>
> error_reporting = E_ALL & ~E_NOTICE
>
> Karel
>
> Adel Al-Hezmi wrote:
>
>
>>Hello all,
>>
>>
>>
>>Last week I sent a question about my problem with ser_fifo and ser web
>>interface, that the php script could not open ser_fifo for writing, in
>>spite of changing access permission of ser_fifo allowing php scripts for
>>read and write. I parsed the error_log file of http server and found
>>that the oohforms.inc in phplib generates a lot of errors, such as
>>undefined index or property. Here is a part of the error_log file:
>>
>>[
>>
>>client 10.10.11.36] PHP Notice: Undefined offset: 9 in
>>/var/www/html/ser/html/page.php on line 219, referer:
>>
>
> http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d
> 9ac75df3a91a5bc0b1b7c3b0
>
>>[client 10.10.11.36] PHP Notice: Undefined offset: 10 in
>>/var/www/html/ser/html/page.php on line 223, referer:
>>
>
> http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d
> 9ac75df3a91a5bc0b1b7c3b0
>
>>[client 10.10.11.36] PHP Notice: Undefined variable: message in
>>/var/www/html/ser/html/user_interface/my_account.php on line 356,
>>referer:
>>
>
> http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d
> 9ac75df3a91a5bc0b1b7c3b0
>
>>[client 10.10.11.36] PHP Notice: Undefined index: frozen in
>>/var/www/html/ser/phplib/oohforms.inc on line 284, referer:
>>
>
> http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d
> 9ac75df3a91a5bc0b1b7c3b0
>
>>[client 10.10.11.36] PHP Notice: Undefined index: frozen in
>>/var/www/html/ser/phplib/oohforms.inc on line 284, referer:
>>
>
> http://10.10.11.36/ser/html/user_interface/index.php?phplib_Session=783a0f6d
> 9ac75df3a91a5bc0b1b7c3b0
>
>>
>>
>>I hope some can help me J
>>
>>
>>
>>Best regards Adel Al-Hezmi
>>
>>
>>------------------------------------------------------------------------
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hi,
thanks for your hint. Could you please let me know what you mean with
SER multihomed will not work anymore without forking.
Does multihomed mean one SER is listening to more than one IP addresses?
This is not the case in my scenario.
Does forking only influence the multihomed "mode" or does it also affect
the SER performance or call processing behavior
(e.g. parallel call attempts are being serialized without forking)??
Regards,
Urs Uschmann
You can use fork=no in ser.cfg and it will allow supervise to control
it, the only caveat is that SER can no longer be multihomed when it
doesn't fork. Also since SER only listens on the one interface you need
to make sure it's listening on the correct interface by passing the -l
flag in your run script in the /service directory.
-Evan
On Thu, 2005-01-27 at 18:49 +0100, Jan Janak wrote:
> We have been using tool called monit:
>
> http://www.tildeslash.com/monit/
>
> Jan.
>
> On 21-01 15:36, u2 wrote:
> > <http://cr.yp.to/djb.html>Hello,
> >
> > did somebody success to control a SER proxy with D.J. Bernstein's
> > daemontools?
> > As soon as SER is run, it puts itself to the shell background. As
far as
> > I found out, this behavior prevents the daemontools from
controlling SER.
> > What did anybody already experiment with this toolkit? What do you use
> > to control you SER processes (auto-respawn, logging, etc.)?
> >
> > It would be also great to use Bernstein's multilog to collect SER logs
> > in a comfortable manner (with per instance log, log rotation,
> > timestamps, etc).
> > What do you use here? Any recommendation?
Ashling O'Driscoll schrieb:
> Hi,
>
> I have the exact same problem as described in the below email which I
> found on the serusers mailing archives. That is clients behind the
> same nat can ring each other and transmit audio but if a client tries
> to ring a client behind another nat no audio is transmitted.
> According to the below email this is because the sdp is not rewritten
> correctly. The below solution mentions calling force_rtpproxy but
> that function is called in the original config (see below)...Is there
> a specific plce that it should be invoked?
>
> Thank you,
> Aisling.
>
> You have to call force_rtpproxy -> this will rewrite the IP address
> and port with the one of the rtpproxy.
>
> klaus
>
> Sun Yen-Rong (Dan) wrote:
>
>
>
>> Hi Klaus,
>> Firstly, thanks for your help. I used ethereal to capture the SIP
>> messages, and found out that the SDP in the forwarded INIVITE and
>>
>
> 200 OK
>
>
>> message are not rewritten correctly. As a result, the client with a
>> public IP address is sending RTP packets to the private IP address
>>
>
> of
>
>
>> the other client who is behind a NAT. One seruser suggested me to
>>
>
> use
>
>
>> fix_nated_sdp("3") instead of fix_nated_sdp("1"), however, it didn't
>> solve my problem. Do I also need to add a parameter to other
>>
>
> nathelper
>
>
>> functions as well?
>>
>> Regards,
>> Dan.
>>
>>
>> -----Original Message-----
>> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at] Sent:
>> Thursday, May 06, 2004 9:45 PM
>> To: Sun Yen-Rong (Dan)
>> Cc: serusers at iptel.org
>> Subject: Re: [Serusers] no audio problem if clients behind NATs
>>
>> use a packet sniffer (ethereal or ngrep) and watch the SIP messages
>>
>
> at
>
>> the SIP proxy. Take a look at the sdp in the forwarded INVITE and
>>
>
> 200 OK
>
>
>> message and verify that the IP address and port in the SDP are
>>
>
> rewritten
>
>
>> correctly and points to the rtpproxy.
>>
>> regards,
>> klaus
>>
>> Sun Yen-Rong (Dan) wrote:
>>
>>
>>
>>
>>> Hi all,
>>>
>>> I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to
>>>
>>
>> make
>>
>>
>>
>>> a call between two clients which are behind the same NAT, everything
>>> work fine. However, when I try to make a call between clients which
>>>
>>
>> are
>>
>>
>>
>>> behind different NATs, niether client can hear each other's audio.
>>>
>>
> my
>
>
>>> configuration file is shown in the end of this message. Can someone
>>>
>>
>> help
>>
>>
>>
>>> me, please?
>>>
>>> Thanks,
>>>
>>> Dan.
>>>
>>> #
>>> # $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $
>>> #
>>> # simple quick-start config script including nathelper support
>>>
>>> # This default script includes nathelper support. To make it work
>>> # you will also have to install Maxim's RTP proxy. The proxy is
>>>
>>
>> enforced
>>
>>
>>
>>> # if one of the parties is behind a NAT.
>>> #
>>> # If you have an endpoing in the public internet which is known to
>>> # support symmetric RTP (Cisco PSTN gateway or voicemail, for
>>>
>>
>> example),
>>
>>
>>
>>> # then you don't have to force RTP proxy. If you don't want to
>>>
>>
> enforce
>
>
>>> # RTP proxy for some destinations than simply use t_relay() instead
>>>
>>
> of
>
>
>>> # route(1)
>>> #
>>> # Sections marked with !! Nathelper contain modifications for
>>>
>>
>> nathelper
>>
>>
>>
>>> #
>>> # NOTE !! This config is EXPERIMENTAL !
>>> #
>>> # ----------- global configuration parameters
>>>
>>
> ------------------------
>
>
>>> debug=3 # debug level (cmd line: -dddddddddd)
>>> fork=yes
>>> log_stderror=no # (cmd line: -E)
>>>
>>> check_via=no # (cmd. line: -v)
>>> dns=no # (cmd. line: -r)
>>> rev_dns=no # (cmd. line: -R)
>>> port=5060
>>> children=4
>>> fifo="/tmp/ser_fifo"
>>>
>>> # ------------------ module loading
>>>
>>
> ----------------------------------
>
>
>>> ",#
>>> # $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $
>>> #
>>> # example script showing use of nathelper module # (incomplete for
>>> sake of brevity)
>>> #
>>>
>>> # ----------- global configuration parameters
>>>
>>
> ------------------------
>
>
>>> # ------------------ module loading
>>>
>>
> ----------------------------------
>
>
>>> loadmodule "/usr/local/lib/ser/modules/sl.so"
>>> loadmodule "/usr/local/lib/ser/modules/tm.so"
>>> loadmodule "/usr/local/lib/ser/modules/rr.so"
>>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
>>> loadmodule "/usr/local/lib/ser/modules/usrloc.so"
>>> loadmodule "/usr/local/lib/ser/modules/registrar.so"
>>> loadmodule "/usr/local/lib/ser/modules/textops.so"
>>>
>>> # !! Nathelper
>>> loadmodule "/usr/local/lib/ser/modules/nathelper.so"
>>>
>>> # ----------------- setting module-specific parameters
>>>
>>
> ---------------
>
>
>>> # -- usrloc params --
>>>
>>> modparam("usrloc", "db_mode", 0)
>>>
>>> # -- rr params --
>>> # add value to ;lr param to make some broken UAs happy
>>> modparam("rr", "enable_full_lr", 1)
>>>
>>> # !! Nathelper
>>> modparam("registrar", "nat_flag", 6)
>>> modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
>>> modparam("nathelper", "ping_nated_only", 1) # Ping only clients
>>>
>>
>> behind
>>
>>
>>
>>> NAT
>>>
>>> # ------------------------- request routing logic
>>>
>>
> -------------------
>
>
>>> # main routing logic
>>>
>>> route{
>>>
>>> # initial sanity checks -- messages with
>>> # max_forwards==0, or excessively long requests
>>> if (!mf_process_maxfwd_header("10")) {
>>> sl_send_reply("483","Too Many Hops");
>>> break;
>>> };
>>> if (msg:len >= max_len ) {
>>> sl_send_reply("513", "Message too big");
>>> break;
>>> };
>>>
>>> # !! Nathelper
>>> # Special handling for NATed clients; first, NAT test is
>>> # executed: it looks for via!=received and RFC1918 addresses
>>> # in Contact (may fail if line-folding is used); also,
>>> # the received test should, if completed, should check all
>>> # vias for rpesence of received
>>> if (nat_uac_test("3")) {
>>> # Allow RR-ed requests, as these may indicate that
>>> # a NAT-enabled proxy takes care of it; unless it is
>>> # a REGISTER
>>>
>>> if (method == "REGISTER" || ! search("^Record-Route:"))
>>>
>>
>> {
>>
>>
>>
>>> log("LOG: Someone trying to register from private
>>>
>>
>> IP,
>>
>>
>>
>>> rewriting\n");
>>>
>>> # This will work only for user agents that support
>>>
>>
>> symmetric
>>
>>
>>
>>> # communication. We tested quite many of them and
>>>
>>
>> majority is
>>
>>
>>
>>> # smart enough to be symmetric. In some phones it
>>>
>>
>> takes a
>>
>>
>>
>>> configuration
>>> # option. With Cisco 7960, it is called
>>>
>>
>> NAT_Enable=Yes, with
>>
>>
>>
>>> kphone it is
>>> # called "symmetric media" and "symmetric
>>>
>>
>> signalling".
>>
>>
>>
>>> fix_nated_contact(); # Rewrite contact with source
>>>
>>
>> IP of
>>
>>
>>
>>> signalling
>>> if (method == "INVITE") {
>>> fix_nated_sdp("1"); # Add direction=active to
>>>
>>
>> SDP
>>
>>
>>
>>> };
>>> force_rport(); # Add rport parameter to topmost Via
>>> setflag(6); # Mark as NATed
>>> };
>>> };
>>>
>>> # we record-route all messages -- to make sure that
>>> # subsequent messages will go through our proxy; that's
>>> # particularly good if upstream and downstream entities
>>> # use different transport protocol
>>> if (!method=="REGISTER") record_route();
>>>
>>> # subsequent messages withing a dialog should take the
>>> # path determined by record-routing
>>> if (loose_route()) {
>>> # mark routing logic in request
>>> append_hf("P-hint: rr-enforced\r\n"); route(1);
>>> break;
>>> };
>>>
>>> if (!uri==myself) {
>>> # mark routing logic in request
>>> append_hf("P-hint: outbound\r\n"); route(1);
>>> break;
>>> };
>>>
>>> # if the request is for other domain use UsrLoc
>>> # (in case, it does not work, use the following command
>>> # with proper names and addresses in it)
>>> if (uri==myself) {
>>>
>>> if (method=="REGISTER") {
>>>
>>> # Uncomment this if you want to use digest authentication
>>> # if (!www_authorize("iptel.org", "subscriber")) {
>>> # www_challenge("iptel.org", "0");
>>> # break;
>>> # };
>>>
>>> save("location");
>>> break;
>>> };
>>>
>>> lookup("aliases");
>>> if (!uri==myself) {
>>> append_hf("P-hint: outbound alias\r\n");
>>> route(1);
>>> break;
>>> };
>>>
>>> # native SIP destinations are handled using our USRLOC
>>>
>>
>> DB
>>
>>
>>
>>> if (!lookup("location")) {
>>> sl_send_reply("404", "Not Found");
>>> break;
>>> };
>>> };
>>> append_hf("P-hint: usrloc applied\r\n"); route(1);
>>> }
>>>
>>> route[1] {
>>> # !! Nathelper
>>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
>>>
>>
>> &&
>>
>>
>>
>>> !search("^Route:")){
>>> sl_send_reply("479", "We don't forward to private IP
>>>
>>
>> addresses");
>>
>>
>>
>>> break;
>>> };
>>>
>>> # if client or server know to be behind a NAT, enable relay
>>> if (isflagset(6)) {
>>> force_rtp_proxy();
>>> };
>>>
>>> # NAT processing of replies; apply to all transactions (for
>>>
>>
>> example,
>>
>>
>>
>>> # re-INVITEs from public to private UA are hard to identify as
>>> # NATed at the moment of request processing); look at replies
>>> t_on_reply("1");
>>>
>>> # send it out now; use stateful forwarding as it works reliably
>>> # even for UDP2TCP
>>> if (!t_relay()) {
>>> sl_reply_error();
>>> };
>>> }
>>>
>>> # !! Nathelper
>>> onreply_route[1] {
>>> # NATed transaction ?
>>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>>> fix_nated_contact();
>>> force_rtp_proxy();
>>> # otherwise, is it a transaction behind a NAT and we did not
>>> # know at time of request processing ? (RFC1918 contacts)
>>> } else if (nat_uac_test("1")) {
>>> fix_nated_contact();
>>> };
>>> }
>>>
>>
>
>
>
> -------------------Legal
> Disclaimer---------------------------------------
>
> The above electronic mail transmission is confidential and intended
> only for the person to whom it is addressed. Its contents may be
> protected by legal and/or professional privilege. Should it be
> received by you in error please contact the sender at the above quoted
> email address. Any unauthorised form of reproduction of this message
> is strictly prohibited. The Institute does not guarantee the security
> of any information electronically transmitted and is not liable if the
> information contained in this communication is not a proper and
> complete record of the message as transmitted by the sender nor for
> any delay in its receipt.
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
If it is not symmetric NAT, try to ask a STUN Server outside in the
internet. You just need a client which supports STUN (and it works)! The
clients learn about the NAT (kind of NAT and outbound address) and
change their SDP therefore. A good software choice is XTen lite. A STUN
Server is at sip.technikum-wien.at (Ports 3478, 3479)
regards,
Philipp
Hi,
I have the exact same problem as described in the below email which I
found on the serusers mailing archives. That is clients behind the
same nat can ring each other and transmit audio but if a client tries
to ring a client behind another nat no audio is transmitted.
According to the below email this is because the sdp is not rewritten
correctly. The below solution mentions calling force_rtpproxy but
that function is called in the original config (see below)...Is there
a specific plce that it should be invoked?
Thank you,
Aisling.
You have to call force_rtpproxy -> this will rewrite the IP address
and
port with the one of the rtpproxy.
klaus
Sun Yen-Rong (Dan) wrote:
> Hi Klaus,
> Firstly, thanks for your help. I used ethereal to capture the SIP
> messages, and found out that the SDP in the forwarded INIVITE and
200 OK
> message are not rewritten correctly. As a result, the client with a
> public IP address is sending RTP packets to the private IP address
of
> the other client who is behind a NAT. One seruser suggested me to
use
> fix_nated_sdp("3") instead of fix_nated_sdp("1"), however, it didn't
> solve my problem. Do I also need to add a parameter to other
nathelper
> functions as well?
>
> Regards,
> Dan.
>
>
> -----Original Message-----
> From: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> Sent: Thursday, May 06, 2004 9:45 PM
> To: Sun Yen-Rong (Dan)
> Cc: serusers at iptel.org
> Subject: Re: [Serusers] no audio problem if clients behind NATs
>
> use a packet sniffer (ethereal or ngrep) and watch the SIP messages
at
> the SIP proxy. Take a look at the sdp in the forwarded INVITE and
200 OK
>
> message and verify that the IP address and port in the SDP are
rewritten
>
> correctly and points to the rtpproxy.
>
> regards,
> klaus
>
> Sun Yen-Rong (Dan) wrote:
>
>
>>Hi all,
>>
>>I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to
>
> make
>
>>a call between two clients which are behind the same NAT, everything
>>work fine. However, when I try to make a call between clients which
>
> are
>
>>behind different NATs, niether client can hear each other's audio.
my
>>configuration file is shown in the end of this message. Can someone
>
> help
>
>>me, please?
>>
>>Thanks,
>>
>>Dan.
>>
>>#
>># $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $
>>#
>># simple quick-start config script including nathelper support
>>
>># This default script includes nathelper support. To make it work
>># you will also have to install Maxim's RTP proxy. The proxy is
>
> enforced
>
>># if one of the parties is behind a NAT.
>>#
>># If you have an endpoing in the public internet which is known to
>># support symmetric RTP (Cisco PSTN gateway or voicemail, for
>
> example),
>
>># then you don't have to force RTP proxy. If you don't want to
enforce
>># RTP proxy for some destinations than simply use t_relay() instead
of
>># route(1)
>>#
>># Sections marked with !! Nathelper contain modifications for
>
> nathelper
>
>>#
>># NOTE !! This config is EXPERIMENTAL !
>>#
>># ----------- global configuration parameters
------------------------
>>
>>debug=3 # debug level (cmd line: -dddddddddd)
>>fork=yes
>>log_stderror=no # (cmd line: -E)
>>
>>check_via=no # (cmd. line: -v)
>>dns=no # (cmd. line: -r)
>>rev_dns=no # (cmd. line: -R)
>>port=5060
>>children=4
>>fifo="/tmp/ser_fifo"
>>
>># ------------------ module loading
----------------------------------
>>",#
>># $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $
>>#
>># example script showing use of nathelper module
>># (incomplete for sake of brevity)
>>#
>>
>># ----------- global configuration parameters
------------------------
>>
>># ------------------ module loading
----------------------------------
>>loadmodule "/usr/local/lib/ser/modules/sl.so"
>>loadmodule "/usr/local/lib/ser/modules/tm.so"
>>loadmodule "/usr/local/lib/ser/modules/rr.so"
>>loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
>>loadmodule "/usr/local/lib/ser/modules/usrloc.so"
>>loadmodule "/usr/local/lib/ser/modules/registrar.so"
>>loadmodule "/usr/local/lib/ser/modules/textops.so"
>>
>># !! Nathelper
>>loadmodule "/usr/local/lib/ser/modules/nathelper.so"
>>
>># ----------------- setting module-specific parameters
---------------
>>
>># -- usrloc params --
>>
>>modparam("usrloc", "db_mode", 0)
>>
>># -- rr params --
>># add value to ;lr param to make some broken UAs happy
>>modparam("rr", "enable_full_lr", 1)
>>
>># !! Nathelper
>>modparam("registrar", "nat_flag", 6)
>>modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
>>modparam("nathelper", "ping_nated_only", 1) # Ping only clients
>
> behind
>
>>NAT
>>
>># ------------------------- request routing logic
-------------------
>>
>># main routing logic
>>
>>route{
>>
>> # initial sanity checks -- messages with
>> # max_forwards==0, or excessively long requests
>> if (!mf_process_maxfwd_header("10")) {
>> sl_send_reply("483","Too Many Hops");
>> break;
>> };
>> if (msg:len >= max_len ) {
>> sl_send_reply("513", "Message too big");
>> break;
>> };
>>
>> # !! Nathelper
>> # Special handling for NATed clients; first, NAT test is
>> # executed: it looks for via!=received and RFC1918 addresses
>> # in Contact (may fail if line-folding is used); also,
>> # the received test should, if completed, should check all
>> # vias for rpesence of received
>> if (nat_uac_test("3")) {
>> # Allow RR-ed requests, as these may indicate that
>> # a NAT-enabled proxy takes care of it; unless it is
>> # a REGISTER
>>
>> if (method == "REGISTER" || ! search("^Record-Route:"))
>
> {
>
>> log("LOG: Someone trying to register from private
>
> IP,
>
>>rewriting\n");
>>
>> # This will work only for user agents that support
>
> symmetric
>
>> # communication. We tested quite many of them and
>
> majority is
>
>> # smart enough to be symmetric. In some phones it
>
> takes a
>
>>configuration
>> # option. With Cisco 7960, it is called
>
> NAT_Enable=Yes, with
>
>>kphone it is
>> # called "symmetric media" and "symmetric
>
> signalling".
>
>> fix_nated_contact(); # Rewrite contact with source
>
> IP of
>
>>signalling
>> if (method == "INVITE") {
>> fix_nated_sdp("1"); # Add direction=active to
>
> SDP
>
>> };
>> force_rport(); # Add rport parameter to topmost Via
>> setflag(6); # Mark as NATed
>> };
>> };
>>
>> # we record-route all messages -- to make sure that
>> # subsequent messages will go through our proxy; that's
>> # particularly good if upstream and downstream entities
>> # use different transport protocol
>> if (!method=="REGISTER") record_route();
>>
>> # subsequent messages withing a dialog should take the
>> # path determined by record-routing
>> if (loose_route()) {
>> # mark routing logic in request
>> append_hf("P-hint: rr-enforced\r\n");
>> route(1);
>> break;
>> };
>>
>> if (!uri==myself) {
>> # mark routing logic in request
>> append_hf("P-hint: outbound\r\n");
>> route(1);
>> break;
>> };
>>
>> # if the request is for other domain use UsrLoc
>> # (in case, it does not work, use the following command
>> # with proper names and addresses in it)
>> if (uri==myself) {
>>
>> if (method=="REGISTER") {
>>
>># Uncomment this if you want to use digest authentication
>># if (!www_authorize("iptel.org", "subscriber")) {
>># www_challenge("iptel.org", "0");
>># break;
>># };
>>
>> save("location");
>> break;
>> };
>>
>> lookup("aliases");
>> if (!uri==myself) {
>> append_hf("P-hint: outbound alias\r\n");
>> route(1);
>> break;
>> };
>>
>> # native SIP destinations are handled using our USRLOC
>
> DB
>
>> if (!lookup("location")) {
>> sl_send_reply("404", "Not Found");
>> break;
>> };
>> };
>> append_hf("P-hint: usrloc applied\r\n");
>> route(1);
>>}
>>
>>route[1]
>>{
>> # !! Nathelper
>> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
>
> &&
>
>>!search("^Route:")){
>> sl_send_reply("479", "We don't forward to private IP
>
> addresses");
>
>> break;
>> };
>>
>> # if client or server know to be behind a NAT, enable relay
>> if (isflagset(6)) {
>> force_rtp_proxy();
>> };
>>
>> # NAT processing of replies; apply to all transactions (for
>
> example,
>
>> # re-INVITEs from public to private UA are hard to identify as
>> # NATed at the moment of request processing); look at replies
>> t_on_reply("1");
>>
>> # send it out now; use stateful forwarding as it works reliably
>> # even for UDP2TCP
>> if (!t_relay()) {
>> sl_reply_error();
>> };
>>}
>>
>># !! Nathelper
>>onreply_route[1] {
>> # NATed transaction ?
>> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
>> fix_nated_contact();
>> force_rtp_proxy();
>> # otherwise, is it a transaction behind a NAT and we did not
>> # know at time of request processing ? (RFC1918 contacts)
>> } else if (nat_uac_test("1")) {
>> fix_nated_contact();
>> };
>>}
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hi to all
i hope that this message is not OT here .. i'm looking for DID in France ... is there someone here that provides this service? (well if not it's great also do have suggestions about where to find it)
thnx a lot for ur support
Graziano