Hi,
I have tried to verify if ser supports multihomed computers without any luck.
Has anybody and hint, pointer to where to find info about this topic?
Kind regrads
Mats
Hi Serusers,
I have a slight problem with aliases.
If I add the alias directly to "aliases" table in the DB, it never gets
reloaded into SER memory.
If I add the alias through the FIFO interface, the "username" property
doesnt get filled out correctly. In my "location" it is
<username>@<domain> but when adding to "aliases" throught the FIFO I only
get <username> (without @<domain>) .. I suppose I need it to be the same
in aliases table for lookup("aliases") to work correctly, atleast its not
working when its only the username without username@domain there.
Anyone got some suggestions for what to do ?
Thanks, Soren
--
Soren Davidsen,
Univ: AUE department of computer science, room H302, +45-20676081
Home: http://tanesha.net/, +45-36947805 or +1-3602266324
Ring billigt med IP-telefoni: http://musimi.dk/, http://GratisSip.dk/
hello friends
extremly soory for previous mail ,by mistake i send
that
i am trying to establish a call between two private
ips A 215104,B 214103 both can able to register but it
i could able
to call from only one private ip phone A to another
private ip phone B not in vice versa .
voice could be able to listen in both sides
in ethereal report what i got is
i have seen clearly that 215104 is registering
but when 214103 is calling it server is trying to
get 215104 but it 215104 gives the message backto the
server as Icmp unreachable.
in debug mode ser is showing this messages while
calling
it shows that 215104 not found in usrloc
ser.cfg is like this
*******************************************************
#
# $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=8 # debug level (cmd line:
-dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
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") {
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();
};
}
*******************************************************
with regards
serdie hard
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hello friends
i am trying to establish a call between two private
ips A,B both can able to register but it i could able
to call from only one private ip phone A to another
private ip phone B not in vice versa .
voice could be able to listen in both sides
in ethereal report what i got is
****************************************************
******************************************************
in debug mode ser is showing this messages while
calling
from one 214103(a)202.*.*.19 to 215104(a)202.*.*.19
******************************************************
*******************************************************y
ser.cfg is like this
*******************************************************
*******************************************************
__________________________________
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Hi LIST
I am trying to work on Call waiting but unable to make a start. Can anybody
guide me....
Regards
_________________________________________________________________
Get FREE unlimited Citibank ATM transactions!
http://go.msnserver.com/IN/54174.asp Discover real-value banking here.
At 09:09 AM 8/17/2004, Soren (Home) wrote:
>Try using revert_uri() before you send to voicemail.
>
>It would be great though if sems could be configured to use To: header
>instead of userpart of request-uri.
I don't see it would be so great. Consider call forwarding: user(a)foo.com
sets up forwarding to user(a)bar.com which gets processed by voicemail.
bar.com has no knowledge of what it could do with user(a)foo.com.
-jiri
>Cheers, Soren
>
>> Hey everybody.
>>
>> Thanks for you help on my last voice mail error. I'm having another issue
>> though. I'm using aliases on my subscribers. If another subscriber dials
>> the
>> alias number, they don't get redirected to voicemail. They have to dial
>> the
>> subscribers "normal" number to get proper redirection to voicemail.
>>
>> What am I missing here? I do a lookup("aliases"); and calls terminate. I'm
>> guessing I'm missing a step in there having to rewrite the user portion of
>> the uri?
>>
>> Thanks in advance for your help.
>>
>> Cheers,
>>
>> CJ
>> cto(a)livevoip.com
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>--
>Soren Davidsen,
>Univ: AUE department of computer science, room H302, +45-20676081
>Home: http://tanesha.net/, +45-36947805 or +1-3602266324
>Ring billigt med IP-telefoni: http://musimi.dk/, http://GratisSip.dk/
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi List,
My ser.cfg file has grown quite large now and it's becoming quite hard to
manage all the gateways. For example, if we need to change them.
What I'd like to have is a file that contains the hosts, e.g.
@GW1@ gateway1.mydomin.com
@GW2@ gateway2.mydomin.com
then have a ser.cfg.template script where I'd define the hosts in the
forward/reply commands as @GW1@ etc.. and run a shell script/command on that
file to generate the master ser.cfg file.
My question is that is there already a program available that would allow me
to do this kind of thing?
Regards,
Alan
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Hey All,
How goes? Just a quick one for everyone. Can someone recommend a good
billing style package to use with SER? I'm not looking for the best one on
the market, just one that works best with SER.
Thanks
CJ
Dear All,
I need your help to send the voicemail as an aatchements in GSM file format.
in side the SEMS.conf there is a line like that "sets the file extension which will be used
to record messages" I want to record my voice in GSM file format and i want to send this file as an attachement in GSM file format also.
Could any body advice me how can i do this one.
thanks in advance..
koyama
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I've configured ser to listen on two ports, in the ser.cfg:
listen=a.b.c.d:5060
listen=a.b.c.d:5062
The idea is that 'normal' public traffic and traffic that can be handled by
normal 'nat' detectors will be directed to port 5060, but traffic that I
know is difficult is directed to register and use a.b.c.d:5062 as the SIP
server.
This all works great in the NAT (inside) -> SER -> somewhere direction, but
I've discovered that when any device which registers to the 5062 port, the
fact that it arrived on an alternative port doesnt seem to be remembered.
Therefore, when I have a somewhere -> SER -> NAT (inside) call, (on a 5062
registered device) although SER remembers the port on the NAT device to
forward the INVITE to, it sends it from port 5060 - which at least on my
test nat router is enough for it to reject the packet and not pass it
inside.
Any thoughts? Is this a configuration item I've missed, or is this a current
issue with SER?
Linus
(* The other reason for using port 5062 is that it 'SIP aware' but possibily
broken NAT routers won't rewrite the traffic if its not on port 5060)
