Hi All,
I'm new to SER. I have some problems about login the user interface. When login, it seems the login process holds and don't have any response. And the permission of fifo file is set to 666. Could anyone help me to solve this problem ?
Thank you very much.
Regards,
Tracy
Hi,
When a new call comes, ser will forward the INVITE to the UA's location
after lookup(). If the UA is not available, after a few retries, the call
fails. Is there a knob to control how many retries? Currently it retries
after 0.5s, 2.0s, 4.0s, 4.0s, 4.0s, 4.0s. During this 18.5s, there is no
progress indication of the caller. It is too long for most users and most
networks. Is it something we can control in a ser config? If not, which
source file should we look into?
Thanks,
Richard
See CDRTool it has PSTN and NGN rating capability
http://www.ag-projects.com/CDRTool.html
Adrian
> Message: 3
> Date: Sun, 19 Dec 2004 03:44:17 +0800
> From: "Ronald Ramos" <ron(a)silverbackasp.com>
> Subject: [Serusers] country codes
> To: serusers(a)lists.iptel.org
> Message-ID: <20041218194417.M25068(a)silverbackasp.com>
> Content-Type: text/plain; charset=iso-8859-1
>
>
>
> Hi All!
>
> I'm having quite a problem on my billing system.
> Every call, going to any countries has a fixed price, but my gateway
> provider, doesn't charge me fixed, it varies on the country code
> they're
> calling.
>
> What should I do so that I can capture a certain country code and its
> cost.
Hi
This is a cross post from asterisk list...
I was looking at the option of call forwarding and call blasting, etc,and
was wondering whether to do this in ser or in asterisk.
Obviously I would leave the voicemail etc features to asterisk, but the
initial rewrite of the numbers would that be better in SER.
tks
Iqbal
Hi All!
I'm having quite a problem on my billing system.
Every call, going to any countries has a fixed price, but my gateway
provider, doesn't charge me fixed, it varies on the country code they're
calling.
What should I do so that I can capture a certain country code and its cost.
I'm using something like this:
######################################################################
# Route to main Gateway
######################################################################
route[1] {
rewritehost("Main.Gateway.IP.Address");
t_on_failure("1");
t_relay();
}
######################################################################
# Route to secondary Mex. PSTN Gateway
######################################################################
failure_route[1] {
log(1,"Main Gw failed or is busy. Trying secondary.\n");
rewritehost("Second.Gateway.IP.Address");
append_branch();
t_relay();
}
Call will be routed to the second gateway if the first gateway is busy or
down.
Does anybody know a method for load balancong?
Regards,
Jesus
-----Original Message-----
From: Vitor Brasileiro [mailto:vitorbrasileiromail@yahoo.com.br]
Sent: Viernes, 17 de Diciembre de 2004 06:27 p.m.
To: serusers(a)lists.iptel.org
Subject: [Serusers] Backup GW
Hi list,
My SER sends to a specific GW all the external calls, but I intend to have a
secondary (backup) GW to terminate those calls would't be completed for some
reason.
Does nbody knows how?
Thanks,
Vitor Brasileiro.
Abid A. Mirza
Paramount Impex(L.L.C)
#306 Gargash Centre
Deira Dubai, UAE
Tel +9714 223 3218
Fax +9714 223 3713
Mob +97150 551 5013
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/04
Can someone please provide me some hints/answers to
the following questions. I had posted this a few days
ago and have not received any answers...
1) I realize that the SEMS has a conferencing module -
But from what I understand, each user who wants to
participate in a conference would be "dialing" a
specific conferencing URI and they would all be
bridged and conferencing can take place. Is this
correct?
2) A convenient and more realistic scenario is when I
am on a call with someone and I'd like to conference
in someone. Normally, I would put the first user on
hold, Dial a key sequence followed by the phone
number of the person I'd like to conference in. Once
the person picks up, I can bring the other person
on-line and the conference can take place.
What does one require to make this happen? From what I
read, the User Agents have to support this
functionality (mixing?). Can this be done via an ATA
or do I need IP Phones? Which ATA/IP Phones supports
this? Secondly, I assume I'd still require SEMS to act
as a media server? Is this correct? Vonage supports
this feature, so does this imply that the ATA that
they supply provides built-in support?
__________________________________
Do you Yahoo!?
Meet the all-new My Yahoo! - Try it today!
http://my.yahoo.com
Remember to have NAT mapping enabled on the Sipura (Admin > Advanced > Line1), what kind of firewall/router are you using? It must be symmetric, meaning not only does the port 58698 have to remain open but SER must be able to communicate with your device on that same port. This is of course a common problem with NAT's.
I've had problems with linux firewalls running iptables, it would work fine until the UA would reboot, then it would register on another port and for some reason it just wouldn't traverse anymore. Small routers such as linksys, 3com, netgear work just peachy.
-----Original Message-----
From: support [mailto:support@cybertel.biz]
Sent: Wednesday, December 15, 2004 10:09 PM
To: Ricardo Martinez
Cc: serusers(a)lists.iptel.org
Subject: RE: [Serusers] rtpproxy + nathelper problem
Hi,
I use the ser.cfg below. I have rtpproxy running. Sipura 2000 can register now without STUN server. However, the registration port is 58698 not 5060.
Here is the SIP address after registration: sip:811@218.xxx.xxx.xxx:58698
Thomas
# ----------- global configuration parameters ------------------------
debug=3 # 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
*/
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/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
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();
};
}
#-- end of config --
On Wed, 2004-12-15 at 21:39, Ricardo Martinez wrote:
> Hello Thomas.
> I think you're missing a few important things. First i don't see any
> checks for NAT'd endpoints, you need to do something like this:
>
> if (nat_uac_test(1))
> setflag(1);
> force_rport();
> fix_nated_contact();
>
> then in the INVITE:
>
> if (method=="INVITE") {
> record_route();
> if (isflagset(1))
> fix_nated_sdp("3");
> };
> /* set up reply processing */
> t_on_reply("1");
> };
>
> I'm using mediaproxy, so i am not sure about this, maybe somone that's
> using the nathelper module can help you better... you can check the
> example configuration file in the /modules/nathelper directory, that
> could give you some clues.
> +
> Cheers.
>
> Ricardo
>
> -----Mensaje original-----
> De: support [mailto:support@cybertel.biz]
> Enviado el: Miércoles, 15 de Diciembre de 2004 4:38
> Para: serusers(a)lists.iptel.org
> Asunto: [Serusers] rtpproxy + nathelper problem
>
>
> Hi,
>
> I have installed the latest rtpproxy and run it with ser-0.8-14. I
> have also include nathelper modules. When I try to register a siprua
> gateway with SER, the IP is still 192.168.1.xx. It cannot be
> translated to NAT public IP.
>
> Why is that so?
>
> Thanks.
>
>
> Thomas
>
>
> Appendix: my ser.cfg
> --------------------------------------------------
> #
> # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
> #
> # simple quick-start config script
> #
>
> # ----------- global configuration parameters
> ------------------------
>
> debug=3 # debug level (cmd line: -dddddddddd)
> fork=yes
> log_stderror=no # (cmd line: -E)
>
> /* Uncomment these lines to enter debugging mode
> debug=7
> 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"
>
> alias=mydomain.dyndns.org
>
> # ------------------ module loading
> ----------------------------------
>
>
> loadmodule "/usr/local/lib/ser/modules/nathelper.so"
> loadmodule "/usr/local/lib/ser/modules/textops.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"
>
>
> # ----------------- 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)
>
> # ------------------------- 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 (len_gt( max_len )) {
> sl_send_reply("513", "Message too big");
> break;
> };
>
> # 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
> record_route();
> # loose-route processing
> if (loose_route()) {
> t_relay();
> 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") {
> save("location");
> break;
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> break;
> };
> };
>
> #inserted by klaus
> if (method=="INVITE") {
> record_route();
> force_rtp_proxy();
> /* set up reply processing */
> t_on_reply("1");
> };
>
>
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> if (!t_relay()) {
> sl_reply_error();
> };
>
> }
>
> #inserted by klaus
> # all incoming replies for t_onrepli-ed transactions enter here
> onreply_route[1] {
> if (status=~"[12][0-9][0-9]")
> force_rtp_proxy();
> }
>
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>
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