Dear serusers,
can you make me a favor and report to me if you have SER
running somewhere on the Internet and offering public SIP
services?
(Please respond even if you think I am surely aware of your
site and/or maitain a nice relationship with you.)
Thank you!
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hi,
I am a graduate student pursuing Master in Computer Science at University of Kentucky. As part of my Master's project, I am in the process of developing a 'SIPTrace' like software (similar to TCPTrace) that gives a summary of SIP Call Connection Information like
1. Caller Lineno, IP Address
2. Callee Lineno, IP Address
3. Connection : Success/Failure
4. Connection Success: Report
a. Who Hungup the call (who said BYE first)
b. Talktime
c. Codec Used in the call
d. Average Call Setup Time, Termination time
e. Average Call Talktime
5. Connection Failure:
a. Report SIP Response Code and Phrase
6. Longest/Shortest SIP Call Talktime in a group of SIP calls
I would like to know what other important "statistical data" (like the above) I can produce as output. I am interested to produce statistical Information like TCPTrace output. I would like to produce any useful Information at SIP Call Level.
I would appreciate your expertise in SIP. Your Ideas and Suggestion will be of great help to me.
Thanks in Advance,
Balaji
Hello all !
I use ser 0.8.11 with support for i.e abc.net and def.net
two different domains.
I can call from each domain to everywhere
but I'cant call between this domains where
ser is responsible for
call between users registerd in abc.net and def.net
is impossible. Too many hops.
Where and what do I have to put to config to fix it ?
Thanks
Andrzej
All:
Please bear with me... I'm having a hard time formulating this quesiton.
Basics:
ser-0.8.12 from src
rtpproxy = $Id: main.c,v 1.19.2.10 2003/10/18 20:14:22 sobomax Exp
RH 7.3 updated to: [root@sip2 rtpproxy]# uname -a
Linux sip2.claritycolorado.com 2.4.20-20.7 #1 Mon Aug 18 14:56:30 EDT
2003 i686
unknown
The problem seems to be that I cannot "always" forward calls to a given
destination if I have executed the force_rtp_proxy.
Typically I'm using a UA like a Cisco ATA, PureData, or softphone behind
a firewall. They in-turn register to SER/rtpproxy at a legal IP
address; most calls are then forwarded to a Vega gateway that resides on
the same network as SER/rtpproxy (in Denver). In this instance all is
well. However, when I try to alt route calls to an "off-network" source
(Cisco in LA and Sonus in Houston) I get two different and disturbing
results:
1. Cisco will not respond to SER: Cisco simply indicates in its own
logs that the INVITE is invalid and thus "trashed" the message
2. Sonus will respond with a "Bad Request"
Below are some snippets:
Prior to calling the route I have already I identified whether or not my
UA is behind a firewall and have fix_nated_contact(), force_rport(),
setflag(1)
SER/rtpproxy is running at 64.47.72.12
# Clarity Vega Evergreen CO -- this one works
route[1]
{
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;
};
record_route();
if (isflagset(1)) {
fix_nated_sdp("1");
force_rtp_proxy();
t_on_reply("2");
}
rewritehostport("64.47.72.10:5060");
t_relay();
}
# Clarity Cisco AS5300 LA -- this one does not
route[2]
{
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;
};
record_route();
if (isflagset(1)) {
fix_nated_sdp("1");
force_rtp_proxy();
t_on_reply("2");
}
rewritehostport("198.77.133.30:5060");
t_relay();
}
# Clarity Sonus Houston -- this one does not
route[3]
{
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;
};
record_route();
if (isflagset(1)) {
fix_nated_sdp("1");
force_rtp_proxy();
t_on_reply("2");
}
rewritehostport("64.47.6.6:5060");
t_relay();
}
/* inbound */
onreply_route[2] {
fix_nated_contact();
fix_nated_sdp("1");
force_rtp_proxy();
}
Please... If you have any ideas... I'm really stuck.
Jac Barben
Hi,
I'm trying to forward all calls from ser to deltathree's servers, i can
make a call i can CANCEL the call but if the call has started i can't send
BYE to terminate the call. As soon as the conversation starts i get this
message in /var/log/messages
ser[23426]: WARNING: set_timer called on a "detached" timer -- ignoring:
0x283242fc
i'm using ethereal and i can't see BYE request. I'm using :
version: ser 0.8.12 (i386/freebsd)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168 2003/10/12 15:09:08 andrei Exp $
main.c compiled on 12:50:17 Feb 18 2004 with gcc 3.2
and X-Lite 2.0 private build 1101
i'm using the same username and password that i have from deltathree so i
can login to their servers too. here is my config
-------------------------------------------------------------------------------
debug=4
fork=yes
log_stderror=no
listen=192.168.0.2
#listen=127.0.0.1
# hostname matching an alias will satisfy the condition uri==myself".
alias=intranet.com.mk
alias=192.168.0.2
alias=igor.intranet.com.mk
# dns - Uses dns to check if it is necessary to add a "received=" field
# to a via. Default is no.
# rev_dns - Same as dns but use reverse DNS.
dns=no
rev_dns=no
port=5060
children=4
# check_via - Turn on or off Via host checking when forwarding replies.
# Default is no. arcane. looks for discrepancy between name and
# ip address when forwarding replies.
check_via=yes
# syn_branch - Shall the server use stateful synonym branches? It is
# faster but not reboot-safe. Default is yes.
syn_branch=yes
# memlog - Debugging level for final memory statistics report. Default
# is L_DBG -- memory statistics are dumped only if debug is set high.
memlog=3
# sip_warning - Should replies include extensive warnings? By default
# yes, it is good for trouble-shooting.
sip_warning=yes
# fifo - FIFO special file pathname
fifo="/tmp/ser_fifo"
# server_signature - Should locally-generated messages include server's
# signature? By default yes, it is good for trouble-shooting.
server_signature=yes
# reply_to_via - A hint to reply modules whether they should send reply
# to IP advertised in Via. Turned off by default, which means that
# replies are sent to IP address from which requests came.
reply_to_via=no
# user | uid - uid to be used by the server. 99 = nobody.
uid="nobody"
# group | gid - gid to be used by the server. 99 = nobody.
gid="nobody"
# mhomed -- enable calculation of outbound interface; useful on
# multihomed servers.
mhomed=0
# ------------- external module loading
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/print.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
# ------------- tm parameters
modparam("tm", "fr_timer", 30)
modparam("tm", "fr_inv_timer", 120)
# ------------- rr parameters
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)
# ------------- accounting parameters
modparam("acc", "db_url", "sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 3)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 1)
modparam("acc", "early_media", 1)
# Account failed transactions
modparam("acc", "failed_transactions", 1)
modparam("acc", "log_fmt", "miocfst")
# ------------- usrloc parameters
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 10)
modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser")
# ------------- auth parameters
modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# ------------- routing logic
route{
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message too large");
break;
};
/* ********* RR ********************************** */
if (loose_route()) {
t_relay();
break;
};
/* record-route INVITEs -- all subsequent requests must visit us */
if (method=="INVITE") {
record_route();
} else if (method=="REGISTER") {
log("here is a register");
save("location");
break;
};
# now check if it really is a PSTN destination which should be
handled
# by our gateway; if not, and the request is an invitation, drop
it --
# we cannot terminate it in PSTN; relay non-INVITE requests -- it may
# be for example BYEs sent by gateway to call originator
if (!uri=~"sip:9[0-9]*@.*") {
if (method=="INVITE") {
sl_send_reply("403", "Call cannot be served here");
setflag(1);
} else if(method=="BYE") {
setflag(1);
t_relay();
} else if(method=="CANCEL") {
setflag(1);
t_relay();
} else {
t_relay();
};
break;
};
# account completed transactions via syslog
setflag(1);
strip(1);
rewritehostport("natrelay.deltathree.com:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
break;
};
}
hello, i would like to know if any SER admins/developers out there have any
experience with the PROTOS SIP tests from Oulu. we have been running the
suite against our SER proxy, and would like to compare notes with anyone
else with interesting test results.
(for the record, our tests show SER 0.8.12 to be very robust and resistant
to crashes)
scott t,
DSi
Hi all.
I'm testing the very basic functionalities of SER as a proxy with two soft
phones, one for Windows (phone(a)albasoft.com) and one for Linux
(phone2(a)albasoft.com). The proxy is at 192.168.0.15.
There's no problem with phone -> phone2 communication, but no way in
establishing phone2 -> phone: I always get the following message:
404, Not Found.
As I have been able to see with Ethereal, phone2 tries to INVITE
sip:phone@192.168.0.15:5060, and the proxy responds by saying it can't find such
user (but user sip:phone@albasoft.com is indeed registered!). I think the
problem is the :5060 part, but I'm not sure.
Can anybody help me?. Thans in advance.
Oh, yes, my config file (as you will see, it's the simplest):
listen=192.168.0.15
alias=albasoft.com
alias=albasoft.com:5060
alias=192.168.0.15
alias=192.168.0.15:5060
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;
};
# 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]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
Eduard San Anselmo
Have you modified the Makefile from acc module s directory? You need to
uncomment some lines there to enable different types of accounting (SQL,
Radius ...).
.Daniel
On 2/24/2004 1:57 PM, Mike Fallows wrote:
>Tried building from source, but still getting the same error.
>
>Did a make all (which made all the modules), then a make install.
>Interestingly, mysql.so was not installed in the modules directory, so I
>copied it over.
>
>Do I need some other flags on the build?
>
>Mike
>
>-----Original Message-----
>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On
>Behalf Of Daniel-Constantin Mierla
>Sent: 24 February 2004 10:34
>To: Mike Fallows
>Cc: serusers(a)lists.iptel.org
>Subject: Re: [Serusers] Accounting question
>
>
>Hello,
>the acc module from RPM has built in only support for system log
>accounting. You have to compile the module from sources setting the
>proper flags in module s Makefile.
>
>.Daniel
>
>On 2/24/2004 11:20 AM, Mike Fallows wrote:
>
>
>
>>I have installed the ser rpm under Redhat 9 and Mysql, which seems to work
>>very well.
>>
>>I do have a small problem with accounting.
>>
>>When I modparam("acc", "db_url", "mysql://user:password@localhost/ser"), I
>>get the following from syslog:
>>
>>Feb 24 10:18:03 ns-nby-ser1 ser: set_mod_param_regex: parameter <db_url>
>>
>>
>not
>
>
>>found in module <acc>
>>Feb 24 10:18:03 ns-nby-ser1 ser: parse error (79,64-65): Can't set module
>>parameter
>>Feb 24 10:18:03 ns-nby-ser1 ser: ERROR: bad config file (1 errors)
>>Feb 24 10:18:03 ns-nby-ser1 ser: ser startup failed
>>
>>
>>Any ideas?
>>
>>Michael Fallows
>>
>>_______________________________________________
>>Serusers mailing list
>>serusers(a)lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>>
>>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
>
Hi this is where error is coming
mail is going perfect but in the mean time...
(4897) DEBUG: run (AmServer.cpp:165): version= <>
(4897) ERROR: run (AmServer.cpp:194): wrong FIFO Interface version.
this is coming which is stopping my sems and ser combination to stop
(5018) DEBUG: openReplyFifo (AmRequest.cpp:362): flags=800
(5018) DEBUG: openReplyFifo (AmRequest.cpp:364): flags=0
(5018) DEBUG: write_to_fifo (AmRequest.cpp:708): Write to fifo: completed
(5018) DEBUG: getReturnCode (AmRequest.cpp:400): Waiting for Ser to respond
to t
he FIFO request...
(4897) DEBUG: run (AmServer.cpp:165): version= <>
(4897) ERROR: run (AmServer.cpp:194): wrong FIFO Interface version.
(5018) DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200
fifo_t_r
eply succeeded
(5018) DEBUG: getReturnCode (AmRequest.cpp:442): n_cmd.from_uri= <>
(5018) DEBUG: getReturnCode (AmRequest.cpp:445): n_cmd.next_hop= <>
(5018) DEBUG: getReturnCode (AmRequest.cpp:448): n_cmd.route= <>
(5018) DEBUG: sendToFIFO (AmRequest.cpp:216): sendToFifo returns
ret=getReturnCo
de=0
(5018) DEBUG: setRAddr (AmRtpStream.cpp:313): RTP remote address set to
203.196.
128.122:54321
(5018) DEBUG: run (AmSession.cpp:141): Sending Rtp data to
203.196.128.122/54321
(5018) DEBUG: onSessionStart (AnswerMachine.cpp:139): starting sending
announce.
..(/usr/local/lib/sems/audio/default.wav)
>From: "Raphael Coeffic" <rco(a)iptel.org>
>To: "Kapil Dhawan" <sersavvy(a)hotmail.com>
>Subject: Re: [Serusers] SER + SEMS
>Date: Tue, 24 Feb 2004 13:16:52 +0100
>
>Hi!
>
> > CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/sems
> > export CVSROOT
> > cvs co answer_machine
>
>you should use instead 'cvs co -r ser_rel_0_8_12 answer_machine'.
>
> > CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/ser
> > export CVSROOT
> > cvs co -r rel_0_8_12 sip_router
>
>Both command work same if you type the CVSROOT as -d parameter as i wrote
>to
>you. It's then important to type it as ONE line.
>(every thing on the same cmd line:
>'cvs -d:pserver:anonymous@cvs.berlios.de:/cvsroot/sems co -r ser_rel_0_8_12
>answer_machine')
>
>-Raphael.
>
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