Hello everybody!
I'm configuring sip server as my student project and I
have few problems, so I'll appreciate your help:
I have configured ser and serWeb, pstn gateway, mysql,
everything is working fine. How I'm unable to put IM
(instant messages) from serweb in function:
- I have configured ser to accept messages in this
way:
...
fifo = "/tmp/ser_fifo"
fifo_mode = 0777
...
loadmodule "/usr/lib/ser/modules/msilo.so"
...
modparam("msilo", "registrar",
"sip:registrar@rcub.bg.ac.yu")
modparam("msilo", "db_url",
"sql://ser:password@rcub.bg.ac.yu/ser")
...
if (method == "REGISTER") {
if (!save("location")) {
sl_reply_error();
};
m_dump();
break;
};
...
if (method == "MESSAGE") {
if (!t_newtran()) {
sl_reply_error();
break;
};
if (m_store("0")) {
t_reply("202", "Accepted for
Later Delivery");
break;
};
t_reply("503", "Service Unavailable");
break;
};
Problem is that serweb continues to report '408
Request timed out' message. I think that maybe there
is problem with fifo, but I'm not sure. I tried to run
shell script for sending IM (from admin guide) but in
syslog there were several errors, all saying that line
must begin with ':' (for lines: NOTIFY,
sip:originator@foo.bar etc.). Why script doesn't work?
Can you help me solve the problem?
Thank you in advance.
Milivoje
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Hello, i am trying to test radius accounting.
I get the following errors when compiling acc, auth_radius, group_radius,
uri_radius modules:
radiusclient.h: No such file or directory
Do I have to install radiusclient first?
If so the Makefiles of these modules has the line:
LIBS=-L$(LOCALBASE)/lib -lradiusclient
I installed radiusclient-0.4.2 and found radiusclient.h in
/usr/local/src/radiusclient-0.4.2/include/radiusclient.h
what do I have to change to be able to compile these modules?
thanks a lot,
Adrian
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hey folks can I make my forwarding or rewrite decisions based on the
attributes I have in the request.
I want to make the forwarding decision based on the user agent or the
various SDP attributes that are there in the INVITE request. How can I
read and compare all the information I find in the SIP protocol
Like based on
1) User-Agent: Asterisk PBX
2) or based on the media type like text , voice , video
Please if any one can help me and provide me with examples .....
thanks
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
Hey can any one tell me how can I do the following
Call a destination (say destination 1)...... if no one picks up then
after certain rings transfer it to destination 2...if no one picks up
there transfer it to voice mail......if any one has already scripted the
config file for this type of application can you share it with me
thanks
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
Hi list:
I am trying nathelper with rtpproxy in bridge mode , (rtpproxy -l "61.61.61.1/192.168.5.1")
UA1----------------------SER+Nathelprt+rtpproxy+NAT----------------------UA2
192.168.5.5 192.168.5.1 / 61.61.61.1 61.61.61.2
When UA1 call UA2 or UA2 call UA1
the call can be established, and the voice can successfully be relay to each UA
but ser shows some error message ,
ERROR: extract_body: message body has lenght zero
ERROR: force_rtp_proxy2: can't extract body from the message
ERROR: on_reply processing failed
is this correct ??
My config file is the one that comes with the files, with the appropiate changes:
debug=3 # debug level (cmd line: -dddddddddd)
#fork=no
#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"
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"
loadmodule "/usr/lib/ser/modules/nathelper.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
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
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();
};
}
Is something wrong? or it is just a warning message?
Thanks in advance
Jimmy
Hi Harris,
I recommend you read Ser HowTo document first, as this will help you get started fairly quickly.
http://iptel.org/ser/doc/ser-howto/ser-Howto.html
Regards,
Freddy
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Harris Coltrain
Sent: Monday, June 14, 2004 1:22 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] SER Configuration
Hello,
I am brand new to SER and need some assistance in setting up SER as a Proxy Server. The objectives are these.
1.) I need users to be able to register via authentication for one domain with SER via MSN Messenger. SJphone, Pingtel Etc. For user info I want to use the built in MySQL mod,and understand how to add users via "secrtl" but do not fully grasp the complete process to get the server to this point. If anyone could point me in the right direction, I would greatly appreciate it. Things that come to mind:
What needs to be uncommented in the ser.cfg?
What do I need to with DNS(i.e DNS_SRV records)?
Do SER have an IRC#?
Thank you,
Harris Coltrain
Sales and Service
Network Data Solutions
502.212.5183
hcoltrain(a)ndsky.net
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Has anyone used search command. If yes please provide me with the
details on how can I use it or give an example...
Since whenever I am trying search command inside the "if" (as shown in
admin guide) it gives me an error
So can any one help me
I am making a script where I can make the forwarding decision based on
the user agent in the INVITE request.
Quick help really appreciated
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
Hi
I am using rtpproxy + nathelper and it is running fine. My question is can
we run rtpproxy on different machine and SER on another (asking for
curiosity). How can i test it if it is possible.
Regards
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