Suppose this simply scenario: 3 SIP phones A,B,C.
On B is set a redirection (moved temporarily) direct to C.
On C is set a redirection (moved temporarily) direct to B. <?xml:namespace
prefix = o ns = "urn:schemas-microsoft-com:office:office" />
If A calls B and B replies with moved temporarily to C.
A INVITE B
A ---------a B
B “Moved temporarily to C”
A s--------- B
A calls C and B replies with moved temporarily to B.
A INVITE C
A ---------a C
C “Moved temporarily to B”
A s--------- C
A calls B
A INVITE B
A ---------a B
so it generates an endless loop.
How can I detect and avoid this endless loops?
Does the INVITE contain an header field, analogous to the “Max Forwards”,
that I can see to identify a loop and consequently refuse the INVITE?
Thanks for your help,
Daniele
Hi,
Has anyone able to integrate a PABX to talk to a SIP server?
So that I can use their existing phone system, but when they dial outside
their network it will be the SIP server handling it.
Thank you
Ronald
http://www.alpha-broadbusiness.jp/
We have the Cisco Unity software
Cisco Unity 4.0.4 software + SER proxy ( rtpproxy + nathelper + radius ) and
it is running fine.
Hi,
I am testing click to talk function working with SER server like below:
SER server --------------------------------------
| |
Web server(click to talk by using click_to_dial function) |
| make call |
Microsoft IE (working as caller) ----------------> IAD or other soft client (working as callee)
I wonder click to dial function in Serweb can work independency and how to deal with media stream(voice data).
Jimmy
----- Original Message -----
From: "Jiri Kuthan" <jiri(a)iptel.org>
To: "Ezequiel Colombo" <ecolombo(a)arcotel.net>; <serusers(a)iptel.org>
Sent: Saturday, May 15, 2004 5:53 AM
Subject: Re: [Serusers] Click To Dial dialog mismatch
> At 10:13 PM 5/14/2004, Ezequiel Colombo wrote:
> >Hi, i am trying the funcion click_to_dial included in the serweb package and i note that the function stored in the functions.php file as "click_to_dial()" is not coded to send the ACK after the receipt of 200 OK from the client. My client (X-Lite) remain sending the 200OK forever and dont process the REFER message sent by the server.
> >
> >In the 72 page of the SER Admin Guide i see the dialog of click to dial function like this :
> >
> > Example 4-12. Call-Flow for Click-To-Dial Using REFER
> > CTD Caller Callee
> > #1 INVITE
> > ----------------->
> > ...
> > caller answers
> > #2 200
> > <-----------------
> > #3 ACK
> > ----------------->
> > #4 REFER
> > ----------------->
> > #5 202
> > <-----------------
> > #6 BYE
> > ----------------->
> > #7 200
> > <-----------------
> > #8 INVITE
> > ------------------>
> > #9 180 ringing
> > <------------------
> >Are the function included in SERWEB working fine on your systems ???
>
> Yes.
>
> ACK is generated autonomously be SER's TM module.
>
> Perhaps SER logs will help you further.
>
> -jiri
>
> >
> >Thanks
> >Ezequiel Colombo
> >_______________________________________________
> >Serusers mailing list
> >Serusers(a)iptel.org
> >http://mail.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)iptel.org
> http://mail.iptel.org/mailman/listinfo/serusers
Hi All
I have been working to make this work.
I have an Asterisk gateway and a Ser proxy running on to different
servers.
Ser has one Public Ip and a private one. ( on two different NICs)
I have installed RTPProxy on the same server as Ser.
When I make a call from my cell phone I can have a conversion on my SIP
phone. Everything works great.
But when I call from SIP phone it is no sound.
My cell rings I can pick it up but no sound.
Can one of you expert on this mater please help me?
I'm enclosing my ser.cfg
#
# $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=4 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=9
fork=yes
log_stderror=yes
*/
alias="Domain.com"
alias="my.Domain.com"
# alias="192.168.0.100"
# alias="192.168.0.200"
listen="public ip"
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=0777
# ------------------ 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"
loadmodule "/usr/local/lib/ser/modules/nathelper.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"
# ----------------- 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")) {
log(1, "NAT client\n");
record_route();
# 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") {
if (method == "REGISTER" || !search("^Record-Route:")) {
log(1, "LOG: Someone trying to register from private
IP, rewriting\n");
# This will work only for user agents that supportsymmetric
# communication. We tested quite many of them andmajority is
# smart enough to be symmetric. In some phones it takesa 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
IPof signalling
if (method == "INVITE") {
log(1, "NAT -> INVITE\n");
fix_nated_sdp("1"); # Add direction=active
to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
append_to_reply("P-NATed-Caller: Yes\r\n");
};
};
# 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=~"^sip:0[0-9]*@my.Domain.com") {
log(1, "Forwarding to Asteriks\n");
route(1);
# rewritehostport("192.168.0.200:5060");
# append_hf("P-hint: GATEWAY\r\n");
# t_relay_to_udp("192.168.0.200", "5060");
#forward(192.168.0.200,5060);
# Where local asterisk is listening
#t_relay();
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") {
log(1, "Myself -> REGISTER\n");
# 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;
};
};
#inserted by klaus
if (method=="INVITE") {
log(1, "INVITE\n");
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();
};
# append_hf("P-hint: usrloc applied\r\n"); route(1);
}
route[1] {
# !! Nathelper
log(1, "ROUTE[1]\n");
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
IPaddresses");
break;
};
# if client or server know to be behind a NAT, enable
relay
if (isflagset(6)) {
log(1, "Flag is 6 (NAT)\n");
if (!is_present_hf("P-RTP-Proxy")) {
force_rtp_proxy();
append_hf("P-RTP-Proxy: YES\r\n");
};
append_hf("P-NATed-Calee: Yes\r\n");
rewritehostport("192.168.0.200:5060");
append_hf("P-hint: GATEWAY\r\n");
t_relay_to_udp("192.168.0.200", "5060");
break ;
};
# NAT processing of replies; apply to all transactions (forexample,
# 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()) {
log(1, "ROUTE[1] -> sl_reply_error\n");
sl_reply_error();
break;
};
}
# !! Nathelper
onreply_route[1] {
log(1, "onreply_route[1]\n");
if (status=~"[12][0-9][0-9]"){
# force_rtp_proxy();
# if (isflagset(6) && status=~"(183)|2[0-9][0-9]") {
fix_nated_contact();
fix_nated_sdp("1");
force_rtp_proxy();
} else #if(nat_uac_test("1"))
{
fix_nated_contact();
force_rtp_proxy();
};
}
regards
-------------------------------------------------------------
Pressis Consulting DA
Sanjay Duggal
Operative CTO
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.707 / Virus Database: 463 - Release Date: 15.06.2004
Hello Jiri:
I saw the following post with your name assocated with it.I did not see a
resolution to the initial question, that is why is this message being
generated?
[Serusers] failure route and ERROR: t_should_relay: status rewrite by
UAS: stored: 408, received: 487
<http://lists.iptel.org/pipermail/serusers/2003-December/004635.html>
In my case the call is being forwarded to an Octel voice mail system
that is
reachable by going through a Cisco router. Can you add some insight into
what
this message means and why
Jun 17 15:18:15 ser[21652]: ERROR: t_should_relay: status rewrite by
UAS: stored: 408, received: 487
The sections of my config file being touched follow:
# redirect user to vm if not availabile? Flag it now in case of
# rewrite and store it in flag 6 for route block 6
if (is_user_in("Request-URI", "voicemail")) {
t_on_failure("6");
setflag(6);
log(1, "[SER]: Flag for VM redirect successful. \n");
} else {
t_on_failure("1");
log(1, "[SER]: Flag for VM redirect unsuccessful. \n");
};
#
failure_route[6] {
xlog("L_INFO", "\n[SER]: Failure Route block #6 Unavailable user:
Time: [%Tf] Method: <%rm> R-uri: <%ru> Contact Header: <%ct>
From uri <%fu> To < %tu> IP source address <%is> \n\n");
append_branch("sip:86423@130.1.5.3");
append_urihf("CC-Diversion: ", "\r\n");
append_hf("P-hint: OFFLINE-VOICEMAIL\r\n");
t_relay();
}
Getting the error
radclient:No token read where we expected an attribute name
when testing the RADIUS install. Saw in past archieves that it could be a
problem with # and comments, so I removed all of them, any other ideas?
AJ Grinnell
After starting ser, I run ser status and got following message:
'ser dead but subsys locked'
How can I fix this?
I remember in the e-mail archive somewhere metioned that you can clean some files to correct this. I can not find it now.
Gary
I find the reason. The version of SER that I used is branch 0.8.12 from CVS server. But SERWEB is compliant with CVS branch 0.8.13. The file ser_mysql.sh from branch 0.8.13 create table admin_privileges.
So are there a branch of SERWEB support SER 0.8.12 ?
Jimmy
----- Original Message -----
From: "wangji" <wangji(a)bjut.edu.cn>
To: <serusers(a)iptel.org>
Sent: Thursday, June 17, 2004 7:39 PM
Subject: [Serusers] DB Error(table admin_privileges not exist) when usingserweb
> Hi all,
> I got a DB error when I tried to login into serweb(both admin and user_interface).
> The message showing on IE is "DB Error: no such table, file: /var/www/novsky.com/serweb/html/data_layer.php:405".
> I checked mysql's log. It query DB ser, is:
> select priv_name, priv_value from admin_privileges where username = 'admin' and domain = 'iptel.org' and (priv_name = 'change_privileges' or priv_name = 'is_admin');
> That table "admin_privileges" is not exist.
> In DB ser, there are these tables:
> +-----------------------+
> | Tables_in_ser |
> +-----------------------+
> | acc |
> | active_sessions |
> | aliases |
> | config |
> | domain |
> | event |
> | grp |
> | location |
> | missed_calls |
> | pending |
> | phonebook |
> | preferences |
> | reserved |
> | server_monitoring |
> | server_monitoring_agg |
> | silo |
> | subscriber |
> | uri |
> | version |
> +-----------------------+
> And these tables is created by ser_mysql.sh.
> How can I do?
>
>
> Best regards,
>
> Jimmy
>
>
--------------------------------------------------------------------------------
> _______________________________________________
> Serusers mailing list
> Serusers(a)iptel.org
> http://mail.iptel.org/mailman/listinfo/serusers
>