Hey Guys!
First, thanks for a great mailinglist!
How is it possible to either set maximum Reinvites when there is no
"reply" from the device, Im am using RecordRouteing..?
The reson why I want to do this is because If a user is temporary
offline, I want to push him QUICK to a announcement.
Anybody had this scenario before? and got an Idea on how to do this?
I had 2 toughts,
1: Count invites, and set a flag, if f.eks some reply
from phone etc, it will set counter to 0 again, and start looking for
reinvites. This idea died because SER does'nt show me the reinvites it
sends.
2: Use modparam("tm", "retr_timer2", 8), Tho this did'nt work for some
werid reson.
- Atle
Dear Friends,
I am trying to use CPL-C module along with SER 0.8.12.
I have compiled 0.8.12 and patched with tm.patch.
Now when i try to start SER i get given error:
--------------- ERROR ---------------------------------------
0(12455) loading module /usr/local/lib/ser/modules/mysql.so
0(12455) loading module /usr/local/lib/ser/modules/sl.so
0(12455) loading module /usr/local/lib/ser/modules/tm.so
0(12455) ERROR: load_module: could not open module </usr/local/lib/ser/modules/tm.so>: /usr/local/l
ib/ser/modules/tm.so: undefined symbol: run_failure_handlers
-------------- END ERROR ---------------------------
Please help me to solve this problem.
I am in urgency to run CPL with SER.
Any help would be greatly appreciated!!
Thanks,
--
Avkash Chauhan
Dear Sir
I had installed and used the SER for a while....
But I hope that I could use the WEB interface to do some configurations or view some users....
Is this SERWEB doing...?
If this is.....could you please provide me the install procedure for the SERWEB....?
Thanks
Yeah first is solved...
Second is if i have three users 111 222 333...111 called up 222 and they are
talking with the beloew mentioned NAT file (as ser.cfg) if i make a call
from 333 to 111 or 333 to 222.....both parties get RING.......if 333 called
up 111 then 333 and 111 will get ring as if 111 is not busy......
>From: Jiri Kuthan <jiri(a)iptel.org>
>To: "Kapil Dhawan" <sersavvy(a)hotmail.com>, serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Fork
>Date: Tue, 04 May 2004 09:32:01 +0200
>
>At 09:29 AM 5/4/2004, Kapil Dhawan wrote:
> >Hi
> >
> >I am using the stable version of SER and rtpproxy....it is working fine
>as such and i am able to get phones registered.....and can make
>calls....Now....if i use fork=no with the same file....SER is unable to
>process Register Requests...
>
>so set fork=yes
>
> >it does nothin for a register request...it just doesn't respond but for
>fork=yes it works correctly....i know fork is used to fork the request to
>more than one locations..
>
>the config option 'fork' is unrelated to SIP forking, it is a debugging
>mode in which
>children processes are nto forked.
>
>-jiri
>
>
> >Could u pls clarify it....
> >
> >
> >Second...i am using two grandstream phones and i can talk....but if i
>make a call from third phone to any of two..instead of busy tone...i get
>Ring...on the caller as well as callee side....
> >following is my ser.cfg
> >
> >#
> ># $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)
> >
> >/* 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();
> > };
> >}
> >
> >_________________________________________________________________
> >Send flowers in 24 hours!
> >http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping.
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers(a)lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
>
>--
>Jiri Kuthan http://iptel.org/~jiri/
>
_________________________________________________________________
Sports, sports and more sports! Keep up with all thats happening!
http://www.msn.co.in/sports/ Stay connected with MSN Sports!
Hi
I am using the stable version of SER and rtpproxy....it is working fine as
such and i am able to get phones registered.....and can make
calls....Now....if i use fork=no with the same file....SER is unable to
process Register Requests...it does nothin for a register request...it just
doesn't respond but for fork=yes it works correctly....i know fork is used
to fork the request to more than one locations..
Could u pls clarify it....
Second...i am using two grandstream phones and i can talk....but if i make a
call from third phone to any of two..instead of busy tone...i get Ring...on
the caller as well as callee side....
following is my ser.cfg
#
# $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)
/* 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();
};
}
_________________________________________________________________
Send flowers in 24 hours!
http://www.fabmall.com/affiliatehtml/redir/nl7.asp At MSN Shopping.
Hello Guys
Please help me with this issue
I have 1 fxo gateway with each port registerd as a SIP user
IP: 192.168.10.253
Port 1 : 201(a)192.168.10.253
Port 2 : 202(a)192.168.10.253
So on
I have in the other side a Fxs gateways (eg 1 port each)
Gateway 192.168.10.200
Port 1 : 101(a)192.168.10.200
Gateway 192.168.10.210
Port 1 : 102(a)192.168.10.200
I need a "hotline" feature if someone call to the por 2 in the FXO
automatically calls port 1 in gateway 192.168.10.210
And if someone calls to the port 1 in the FXO automatically calls to
192.168.10.200
And also Viceversa
If I pickup the phone automatically the gateway goes directly to the port 1
in FXO and that only port not port 2
BTW, please incluye where to put this block into the ser.cfg file
Regards
HA
Hi all,
I am configuring SER in order to use the SEMS's voicemail
when one of my subscribed users is not avaiable or
when all the INVITEs to their contacts fail (fork).
I'am using the following configuration:
route {
....
if (isflagset(4)) { /*voicemail enabled*/
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
break;
}
}
failure_route[1] {
t_reply("100","Trying -- going to voicemail");
if (method=="INVITE" || method=="REFER") {
log("**************** vm start - begin ******************\n");
log("**************** voicemail ******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm start - end ******************\n");
}
else if (method=="BYE") {
log("**************** vm end - begin ******************\n");
log("**************** voicemail ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm end - end ******************\n");
};
}
...
This works fine when the system calls a user's contact whose username part
matches the subscribed user name.
Unfortunately, if the system tries to reach such a contact at a phone
number (using a gateway) the result is that SEMS reports:
"Error: 404 voicemail: no email address for user 034823213213."
Can you help me ?
Best regards,
Andrea
Hello
I want to test a scenario with SER and Vovida B2BUA, but I'm having some
problems. If someone has tested this before, I would like to ask you the
config files to help me.
I'm having problems when Vovida b2bua sends second INVITE to SER and SER
doesn't find UA2 (which is correctly registered) (step 3).
UA2
^
|
| 4
1 2
UA1 ----> SER ------> B2BUA
<-----
3
Thanks for your help
Curro
Hello all
I have a question about presence xml document in SER.
I would like to know whether this xml is based on any RFC or draft.
I've been reading Presence Information Data Format
<draft-ietf-impp-cpim-pidf-08.txt> and I have found nothing about <atom>
element. Should UA support this element?
Could you give some information?
Thank you very much
Curro
Hi,
I am getting the following error:
ERROR 1017 at line 1: Can't find file: './ser/subscriber.frm' (errno: 13)
when I run:
serctl add axel 123 axel(a)avenue500.com
It was first complaining about /etc/my.cnf at line: 1, and I changed that file to:
------ start file -----
[mysqld]
socket=/var/lib/mysql/mysql.sock
------ end file -----
and now I am getting the above error. Any clues what the error means and what I should do?
Best regards,
Axel
