Axel,
I tried +878102233342019 and it returned nada.
Adrian
> i don't know which domain those guys use, but my phone number in
> the official e164.arpa domain was not found.
I have a simple e164.arpa lookup script on my personal page:
http://nona.net/features/enum/
comments appreciated.
cheers
axelm
hello friends,
is gsm 6.10 is supported in the sip express
router
with regards
ser die hard
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - 100MB free storage!
http://promotions.yahoo.com/new_mail
As a preface, my setup is somewhat unique; if anyone has a working SER
configuration for this scenario, or is willing to help come up with
one, I would be willing to compensate you for your efforts. If you're
interested, please contact me off list for details.
At present, I'm using Asterisk to run a small VoIP network with great
success. My focus is now to look at handling things such as NAT and
scalability/reliability. This where I believe SER will be very
beneficial as it handles NAT and scaling very well.
My desired setup is as follows (the only thing behind a NAT are phones):
Phone(s) --> [NAT] --> SER --> Asterisk(FS) --> Asterisk (PSTN)
All phones behind the NAT have their outbound proxy set to SER and
their regular proxy set to Asterisk(FS). SER then proxies outbound
traffic from the phone to the Asterisk(FS) server acting as the
"feature server" ie. VM, conferencing, etc.
The feature server then routes traffic according to the dialplan in
Asterisk. If the call is destined for the PSTN, the dialplan logic in
Asterisk(FS) sends an INVITE to Asterisk(PSTN) which tells
Asterisk(PSTN) to reINVITE the Phone at the public IP of the NAT and
also tells the phone, via SER, to reINVITE Asterisk(PSTN) at its
public IP. The media streams should then flow between the phone and
Asterisk(PSTN) while both SER and Asterisk(FS) are out of the picture
until the call ends and BYEs are processed.
If a call is destined for another SIP device, the process is the same
except that Asterisk(FS) forwards traffic back to the address of the
phone, which is SER since SER is acting as the proxy for all SIP
devices. SER then forwards the traffic back to the NATted phone as
appropriate.
Amazingly, I already have what I believe to be an SER config which
correctly routes all of the SIP/SDP traffic as desired. My current
issue is that during a call intitated by the NATted phone, the phone
and Asterisk(PSTN) have reinvited and are sending RTP to each other
but the phone behind the NAT cannot hear any of the audio being sent
by Asterisk(PSTN) while the phone on the PSTN can hear audio being
sent by the NATted phone; one way audio. What is really odd about
this is that when the PSTN user first answers the call, approximately
1-2 seconds of audio is heard on the NATted phone from the PSTN user
but then no more is heard for the remainder of the call.
For me, I see this as some sort of RTP issue as the first second of
voice RTP traffic is doing exactly what is expected between the phone
and Asterisk(PTSN). I've verified the SRC and DST of the RTP traffic
with Ethereal. I've also tried force_rtp_proxy and rtpproxy with no
success.
It is my understanding that my desired setup is very similar to that
of others such as FWD (Free World Dialup), et al.
My SER config follows. It is based on an Asterisk+SER configuration
someone posted earlier so please point out anything you see which may
not be correct.
I'll also add that this configuration works flawlessly whenever there
is only a phone and the 2 Asterisk servers; ie. no SER and no NAT.
All reINVITES take place as they should and all media is heard as it
should be.
Thanks in advance!
-Curt
#alias=" mydomian.com "
#Alias="192.168.10.100" #ser
#Alias="192.168.10.120" #Asterisk
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"
fifo_mode=438
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/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"
loadmodule "/usr/lib/ser/modules/acc.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)
# -- acc params --
# set reporting log level
modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 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")) {
if(search("^Contact:(.*)192\.168(.*)")) {
log(1,"We're natted\n");
# 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
save_noreply("location");
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
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_noreply("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
# record_route();
if (isflagset(6)) {
fix_nated_sdp("1");
force_rtp_proxy();
t_on_reply("1");
}
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();
fix_nated_sdp("1");
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")) {
} else if (search("Contact:.*192\.168.*")) {
fix_nated_contact();
};
}
Hi,
we encountered problems when trying to load the uri.so module using SER version 8.12. All other modules that we are using are loading without problems, including mysql.so, auth_db.so, auth.so, acc.so etc; is there anything we are missing? The db_url is set to
modparam("uri", "url_db", "sql://serro:47serro11@localhost/ser")
In version 8.13 we are using uri_db for the check_from() function and we encountered no problems there, but we are very curious to know what's wrong with the uri in version 12.
Note that for version 12 we have used the rpm packages in the folder "latest" for Fedora.
Regards and many thanks,
Charles and Andrew
Hi,
I am trying to enable the missed call logging feature in ser. When I simply
add the following single line
modparam ("acc", "db_url", "sql://ser:heslo@localhost/ser")
in the existing ser.cfg, and restart SER, it returns "bad config file (1
errors)"
I have not problem to use it for other modules, like group, tm, etc.
By the way, the version of SER I am using is 0.8.12.
I guess the module name got changed instead of "acc".
Thanks and Best Regards
Linda
Sorry, this may be obvious, but did you load the module at the beginning
of the file?
Usually something like
loadmodule "/usr/lib/ser/modules/acc.so"
Dave
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Linda Xiao
Sent: 24 July 2004 01:17
To: serusers(a)lists.iptel.org
Subject: [Serusers] what the is the module name of acc.so
Hi,
I am trying to enable the missed call logging feature in ser. When I
simply add the following single line
modparam ("acc", "db_url", "sql://ser:heslo@localhost/ser")
in the existing ser.cfg, and restart SER, it returns "bad config file (1
errors)"
I have not problem to use it for other modules, like group, tm, etc.
By the way, the version of SER I am using is 0.8.12.
I guess the module name got changed instead of "acc".
Thanks and Best Regards
Linda
Dear all,
I am a newbie in SIP please excuse me, for my poor questions.
As I am working to add Voicemail service on my SER proxy. I am very
much confuse about how to use SEMS module with SER.
As according to "SIP Express Router v0.11 --Admin Guide" SEMS is
required to provide voicemail capability in SIP server. As i have downloaded
"sems-0.1.0.tar.gz" and installed in my machine where I have
"ser-0.8.12" as a sip server. after that
And I have changed my ser.cfg scripts to provide voicemail capabilities
and to load vm module available in SER.
Now I get stuck How I have to use SEMS module ( I mean to say that
"ans_machine" available in sems )
what is the difference & similarity between SEMS Voicemail and SER vm
module.
On what basis SEMS Voicemail and SER communicate to each other.
Is their any configuration file is required inside the SEMS like SER
(ex. ser.cfg).
what is the proper step to perform a testing of voice mail using these
modules..
Please excuse me if i have asked something wrong and please teach me
about my queries.
Thanks in advance.
koyama
---------------------------------
Do you Yahoo!?
New and Improved Yahoo! Mail - 100MB free storage!
Hi,
I'm running ser-0.8.11 on a Gentoobox.
When phone1 calls phone2 and hangs up when phone2 is ringing phone2
doesn't stop ringing.
If the person who haves phone2 answers before it stops ringing, the
conversation continues with no problem.
The two phones are Zyxel P2000W.
Do you see what is the problem?
Thanks,
Alexandre
My ser.cfg :
debug=3
fork=no
listen=192.168.1.36
log_stderror=yes
check_via=no
dns=no
rev_dns=no
port=5082
fifo="/tmp/ser_fifo"
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"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (method=="REGISTER") {
save("location");
break;
};
if (method=="INVITE") {
if (uri =~ "sip:1003@*") {
forward(192.168.1.35, 5060);
break;
};
if (uri =~ "sip:1004@*") {
forward(192.168.1.37, 5060);
break;
};
};
}
If this is meant for calls to/from PSTN this it could be possible by
toying with the privacy and screening indicators from Remote-Party-Id.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/
122newft/122t/122t13/ftsipext.htm#wp1027186
Adrian
>>>>>>>>>>>>
ok but using SER, is it really the only way to hide one's number when
calling?
no one else has succeded doing this maybe using some prefix like '#31#'
or so?
i can't believe NOBODY uses this kind of feature..
Enrique-
Juha Heinanen <jh at tutpro.com> escribió:
> ser writes:
>
> > What developpement would be needed to implement the possibility for
> a
> > caller to hide his number when calling via ser ?
>
> if you would like to modify the from field, you would need a b2bua.
> you
> can also try if using Anonymous from field would work in your
> environment.
Leon,
1. Media is normally not accepted by the gateway if is not already
negotiated in the signalling. So nobody would be able to get in the
middle unless they have control of the signaling which you took
provision to protect already. So you are on the safe side unless there
are bugs in your gateway.
2. If you use a media session controller you can enforce more checks in
there and allow media ports at PSTN gateway originating only from that
session controller same as you did for
Mvg,
Adrian
>>>
Does anyone have an answer to this ? It's not really SER or Asterisk
related, but more generic about security for a mediagateway..
Regards,
Leon
On Tue, 2004-07-20 at 10:43, Leon de Rooij wrote:
> Hi again :)
>
> Got one more question about using a mediagateway. Right now I've got
> everything configured that SER relays the call to our mediagateway
> (asterisk) when necessary. The mediagateway is also on a public IP,
but
> only accepting UDP port 5060 connections from the SER proxy. (We use
RP
> (reverse path) filtering on our routers, so the IP address cannot be
> spoofed). Come to think of it, I can additionally also filter on MAC
> address since both machines are in the same LAN..
> I read that a lot of people use an RTP proxy for forwarding the RTP
> traffic to the gateway (which in turn is in a private net).
>
> My question is: Is my setup less secure than using the RTP proxy ? If
> so, why ?
>
> Thanks !
>
> Regards,
>
> Leon
