Try changing usrloc::db_mode to 1 ... write-through.
modparam("usrloc", "db_mode", 1)
For the domain thing ... change use_domain to 1.
Cesc
>>> Charles Wang <lazy.charles(a)gmail.com> 02/17/05 10:00AM >>>
Dear ALL:
I add a user such as "serctl add 1011 1011 aaa(a)sip.xxx.net" (format:
serctl add userid passwd email)
It will insert one record to my MYSQL's "subscriber" table.
But I can not find any information in my another table "aliases".
Because when I wanna look the missed calls from serweb, it always
UNION two tables missed_calls and aliases. So, I think, it should be
some records at this aliases table.
I add records to aliases table manually with command:
"serctl alias add 1011 aaa(a)sip.xxx.net" but the field "domain" is still
empty.
So I login to mysql and "update aliases set domain='sip.xxx.net'; "
I finally see the missed calls information.
unclassified
It is not a good idea to create a user with such more steps(add user,
add alias, change domain). Is it my false in my ser.cfg or my
config.php at serweb?
There is no record at my MYSQL's location table. How can I make it not
empty via correct cfg setting?
My ser.cfg is as below:
#
# $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=8 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
listen=xxx.xxx.xxx.xxx
alias=ser.xxx.net
alias=ser
alias=xxx.xxx.xxx.xxx
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/speeddial.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- nathelper params --
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
# -- db_url params --
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc",
"db_url", "mysql://ser:heslo@localhost/ser")
# -- use_domain params --
modparam("auth_db|group|registrar|speeddial|uri_db|usrloc",
"use_domain", 0)
# -- permissions params --
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
# -- accounting params --
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "log_level", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 0)
# -- avp params --
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
#modparam("avpops", "use_domain", "1")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# To use more than one tables example
#modparam("avpops", "db_scheme",
"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name")
# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 22)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
# ------------------------- 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;
};
if (method=="INVITE" || method=="BYE") {
setflag(1);
};
# !! 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("SER: 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("ser.xxx.net", "subscriber"))
{
www_challenge("ser.xxx.net", "0");
break;
};
save("location");
break;
};
if (method=="INVITE" && uri =~ "^sip:0[0-9]*@") {
append_hf("P-hint: PSTN\r\n");
rewritehostport("xxx.xxx.xxx.xxx:5060");
#t_relay_to_udp("xxx.xxx.xxx.xxx","5060");
#t_relay_to_tcp("xxx.xxx.xxx.xxx","5060");
log(1, "SER: matched ^sip:[^0][0-9]*@ - process
in route(1)\n");
route(1);
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();
};
}
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Dear ALL:
I add a user such as "serctl add 1011 1011 aaa(a)sip.xxx.net" (format:
serctl add userid passwd email)
It will insert one record to my MYSQL's "subscriber" table.
But I can not find any information in my another table "aliases".
Because when I wanna look the missed calls from serweb, it always
UNION two tables missed_calls and aliases. So, I think, it should be
some records at this aliases table.
I add records to aliases table manually with command:
"serctl alias add 1011 aaa(a)sip.xxx.net" but the field "domain" is still empty.
So I login to mysql and "update aliases set domain='sip.xxx.net'; "
I finally see the missed calls information.
It is not a good idea to create a user with such more steps(add user,
add alias, change domain). Is it my false in my ser.cfg or my
config.php at serweb?
There is no record at my MYSQL's location table. How can I make it not
empty via correct cfg setting?
My ser.cfg is as below:
#
# $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=8 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
listen=xxx.xxx.xxx.xxx
alias=ser.xxx.net
alias=ser
alias=xxx.xxx.xxx.xxx
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/speeddial.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- nathelper params --
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
# -- db_url params --
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc",
"db_url", "mysql://ser:heslo@localhost/ser")
# -- use_domain params --
modparam("auth_db|group|registrar|speeddial|uri_db|usrloc", "use_domain", 0)
# -- permissions params --
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
# -- accounting params --
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "log_level", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 0)
# -- avp params --
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
#modparam("avpops", "use_domain", "1")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# To use more than one tables example
#modparam("avpops", "db_scheme",
"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name")
# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 22)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
# ------------------------- 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;
};
if (method=="INVITE" || method=="BYE") {
setflag(1);
};
# !! 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("SER: 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("ser.xxx.net", "subscriber")) {
www_challenge("ser.xxx.net", "0");
break;
};
save("location");
break;
};
if (method=="INVITE" && uri =~ "^sip:0[0-9]*@") {
append_hf("P-hint: PSTN\r\n");
rewritehostport("xxx.xxx.xxx.xxx:5060");
#t_relay_to_udp("xxx.xxx.xxx.xxx","5060");
#t_relay_to_tcp("xxx.xxx.xxx.xxx","5060");
log(1, "SER: matched ^sip:[^0][0-9]*@ - process in route(1)\n");
route(1);
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();
};
}
Hello,
is it possible to add more than one response for one user in the radius
users database? Will the module process this correctly?
Is there a list of all known AVPs available somewhere?
Regards,
Martin Koenig
Hi,
I have found the following message from the mailing list.
I also saw your question on RFC compliance and the Sonus equipment: In order to make Grandstream phones register properly when using STUN behind
symmetric NAT, I had to patch nathelper with the rport != port of received
address check. (I use 0.8.14 and I guess you already have the patch with
the development version). The reason is that Grandstream attempts to
rewrite the address using STUN even though it correctly detects a symmetric
NAT. I have seen that this was introduced in a new firmware not long ago (release notes). This pussles me as sources I have seen claims this to beinvalid behavior (which seems correct to me).
It seems that I am facing the same problem now. Is there any solution to us? Where could I download the nathelper patch as described above?
Thank you.
Best Regards,
Thomas
Hi All,
Can anyone help me or kindly send me a sample part of configuration wherein
I will automatically use mediaproxy/rtpproxy for calls coming from a NATted
UA, going to either PSTN or a local extension on my SIP server, without
setting the outbound proxy on the UA's. Just like a transparent proxy in
squid(sorry if i compared it to squid).
TIA
Regards,
Nhadie
Dear ALL:
I have defined these lines at my ser.cfg.
modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops", "uuid_column", "uuid")
modparam("avpops", "username_column", "username")
modparam("avpops", "domain_column", "domain")
modparam("avpops", "attribute_column", "attribute")
modparam("avpops", "value_column", "value")
modparam("avpops", "type_column", "type")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
Q1: But I don't know what it mean about 700 or 701. Is it a numeric ID
of one UA?
And I have refered to http://www.voice-system.ro/docs/avpops
And find many examples in it.
But I still can't understand what it mean about avp_db_load().
my avp table is named as "usr_preferences" and it is empty before I
use avpops modules.
Q2: Is it should have some initilized records in it?
Otherwise, what will be loaded with avp_db_load()?
For this web page said:
avp_db_load("$ruri/domain","i:/domain_preferences");
- loads all AVPs with ID from 'domain_preferences'
table for domain from RURI
Q3: Is "$ruri" the caller user's URI? and the "domain" is the same?
Is "domain_preferences" table name? Is it equal to my "usr_preferences"?
Or it is just a virtual table in memory?
Thank you for your answer my stupid questions.
Charles
Please ignore my previous email. Here is the correct one. Thank you.
----- Original Message -----
From: support
To: serusers(a)lists.iptel.org
Sent: Thursday, February 17, 2005 11:59 AM
Subject: STUN, rtpproxy problem
Hi,
I am running ser-0.8.14 with rtpproxy and nathelper. I have also enabled STUN on the same server.
My SIP UA supports STUN, but my PSTN gateway does not.
If I don't enable STUN on SIP UA, I can route the call to PSTN gateway successfully using rtpproxy.
If I enable STUN on SIP UA, I cannot route the call out to PSTN gateway. On ringing sound is heard on callie, but no voice passing through.
I use t_relay_to_udp to forward the call to PSTN gateway. It seems that if STUN is supported on SIP UA, the call cannot be routed to PSTN gateway.
Hope someone could fix this problem.
Best wishes,
Thomas
My ser.cfg is as follows:
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- 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)
listen=""
port=5060
#children=4
fifo_mode=0666
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- Nathelper params --
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval
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
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("", "subscriber")) {
www_challenge("", "0");
break;
};
save("location");
break;
};
if (uri=~"^sip:866*") {
log(1, "going to PSTN route\n");
route(2);
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();
break;
};
}
route[2] {
force_rtp_proxy();
t_on_reply("1");
t_relay_to_udp("T1 gateway IP","T1 Gateway UDP port");
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
# Not all 2xx messages have a content body so here we make sure
# out Content-Length > 0 to avoid a parse error
if (!search("^Content-Length:\0")) {
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();
};
}
Hi,
I am running ser-0.8.14 with rtpproxy and nathelper. I have also enabled STUN on the same server.
My SIP UA supports STUN, but my PSTN gateway does not.
If I don't enable STUN on SIP UA, I can route the call to PSTN gateway successfully using rtpproxy.
If I enable STUN on SIP UA, I cannot route the call out to PSTN gateway. On ringing sound is heard on callie, but no voice passing through.
I use t_relay_to_udp to forward the call to PSTN gateway. It seems that if STUN is supported on SIP UA, the call cannot be routed to PSTN gateway.
Hope someone could fix this problem.
Best wishes,
Thomas
My ser.cfg is as follows:
# ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
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 ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/lib/ser/modules/auth.so"
#loadmodule "/usr/lib/ser/modules/auth_db.so"
# !! Nathelper
loadmodule "/usr/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- 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 {
fix_nated_contact();
};
}
Hi Sir,
Thank you for your reply.
I was able to run ser using the configuration below.
How would I know Sir that it's forcing it to use rtpproxy, I was able to
make test calls, but using also the SIP sevrer as the NAT gateway (I don't
have that much equipment to test with :( ) So I think voice will really go
through on each since the SIP server is just a hop away from the UA's.
Also i'd like to force only rtpproxy when only making calls to local
extensions, not going to PSTN. I tried adding this location was save:
if (uri=~"^(sip:)[1-9]...@nhadzter\.com) {
route(1);
break;
};
But I can't make any calls, and when I do ngrep on port 5060, I can see only
INVITE's coming from the caller. It seems that it can't find the CALLEE or
the CALLE doesn't respond.
I'd really appreciate any help you can give me. Thank you so much.
=====================================================
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# We will you flag 6 to mark NATed contacts
modparam("registrar", "nat_flag", 6)
# Enable NAT pinging
modparam("nathelper", "natping_interval", 60)
# Ping only contacts that are known to be
# behind NAT
modparam("nathelper", "ping_nated_only", 1)
):
# 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 used); also,
# the received test should, if complete, should check all
# vias for presence 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 smart enough to be symmetric. In some phones, like
# it takes a configuration option. With Cisco 7960, it is
# called NAT_Enable=Yes, with kphone it is called
# "symmetric media" and "symmetric signaling". (The latter
# not part of public released yet.)
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
};
};
#
# Forcing media relay if necessary
#
route[1] {
if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
break;
};
if (isflagset(6)) {
force_rtp_proxy();
t_on_reply("1");
append_hf("P-Behind-NAT: Yes\r\n");
};
if (!t_relay()) {
sl_reply_error();
break;
};
}
onreply_route[1] {
if (status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
};
}