Hello!
I'm using
sl_reply(4xx…);
drop;
to drop invite requests that doesn't follow some custom logic. Is it possible to log such requests with acc module? It works perfect if 4xx comes from gateway – I have a line in DB, but when I use sl_reply, it's not accounted.
Do I miss something, or it's not possible to use acc in this case?
BR,
Alex.
I'm looking for SER "uac modules", but how to configure ser.cfg to use
it?
Could anyone help me?
Thanks.
>>
Hi to all,
I have deployed SER
on public IP, and i can install a communication betwen two IP Voip
client.
Now i want configure SER to call , from a IP VOIP Client, PSTN
phone number.
I' havent a PSTN gateway but I have a Voip Provider
Account (Eutelia www.eutelia.it)
that require authentication in order
to call PSTN phone.
I'm studing the PSTN configuration reported in SER
getting start (http://siprouter.onsip.org/doc/gettingstarted/ch09.html)
It describe how to use rewritehost("XXX.XXX.XXX.XXX") to specify the
PSTN GATEWAY IP ADDRESS,
but doesn't specify nothing about
authentication.
The question is :
Because i havent a PSTN gateway,
It' is possible to configure SER in order to redirect PSTN call to a
VOIP provider
or It is necessary to use Asterisk between SER and the
Voip Provider ?
Any suggestion about ser configuration would be
appreciate.
Thanks in advance.
I'm working with the gw-pstn.cfg example, and am having a problem with
canceling outbound pstn calls.
The invite step seems to work fine. Function route[3] checks for a
pstn in the uri, and uses route[5] to route outbound calls to the pstn
gateway.
But, if the originating client cancels the call before it gets
connected, the cancel sip message doesn't go through the same route[3]
-> route[5] processing. Instead, the processing ends up falling
through to the end of the main route() function, where
lookup("location") fails, because the pstn in the LHS of the uri
doesn't match any known registered local devices. So, the client gets
a "404" error, and the target pstn keeps ringing.
It seems to me that all messages, not just the invites, need to be
rerouted as they are in route[5], so that they are sent to the pstn
gateway, and aren't processed as if for a local device.
Am I missing something here? Have I not configured everything
correctly?
--
Mark Sidell
Partner
Forte, Inc.
919-942-7068
fax 919-969-2844
www.forteinc.com
Hi, where can I get freeradius-client? I searched the source in freeradius, couldn't find it.
Thanks
SW
----- Original Message ----
From: ram <talk2ram(a)gmail.com>
To: Live Great <livegreat007(a)yahoo.com>
Cc: FreeRadius users mailing list <freeradius-users(a)lists.freeradius.org>; users(a)openser.org
Sent: Friday, September 28, 2007 11:09:17 PM
Subject: Re: [OpenSER-Users] Failed to compile openser with freeradius support
On 9/28/07, Live Great <livegreat007(a)yahoo.com> wrote:
HI,
In FreeBSD 6.2, I got this error when I compiled openser with freeradius support.
../../radius.h:36:32: freeradius-client.h
: No such file or directory
install freeradius client before compiling
ram
Hello every body,
I tryed to install openseradmin. But I have got many problems. I
knew that it must install ruby and rubygem. So I have downloaded :
ruby-1.8.5-p2 and rubygem-0.9.4 no problem for their installation.
But when I tried to do: gem install rails --include-dependencies, i have
got a error message:
ERROR: While executing gem...(Gem: RemoteFetcher::FetchError)
getaddrinfo: Temporary failure in name resolution
(SocketError)
So anybody know what's the matter? Thanks very much!!
Yin
My customer would like to hire someone with OpenSER experience to build a system to be collocated at our facility. The system will need to do basic call routing, handle REFER messages, and do LCR.
-Jonathan
Jonathan Creasy
Senior VOIP Engineer
NetLogic, Inc.
314-266-4000 x109
[cid:image001.gif@01C801E5.02EF39B0]
Hi to all,
I have deployed SER on public IP, and i can install a
communication betwen two IP Voip client.
Now i want configure SER to
call , from a IP VOIP Client, PSTN phone number.
I' havent a PSTN
gateway but I have a Voip Provider Account (Eutelia www.eutelia.it)
that require authentication in order to call PSTN phone.
I'm studing
the PSTN configuration reported in SER getting start (http://siprouter.
onsip.org/doc/gettingstarted/ch09.html)
It describe how to use
rewritehost("XXX.XXX.XXX.XXX") to specify the PSTN GATEWAY IP ADDRESS,
but doesn't specify nothing about authentication.
The question is :
Because i havent a PSTN gateway,
It' is possible to configure SER in
order to redirect PSTN call to a VOIP provider
or It is necessary to
use Asterisk between SER and the Voip Provider ?
Any suggestion about
ser configuration would be appreciate.
Thanks in advance.
Horace
Hi,
I'm interested in using DNS blacklisting to stop transmitting SIP
requests towards UAS endpoints that are non-responsive. I am on
version 1.2, but I can't get it to work properly.
Here is what I am trying to do:
1) I get a message from a UAC that needs to be sent towards 1 of 4
destination UAS endpoints.
2) I setup the OpenSER to perform an SRV lookup which returns a record
that has four A records associated with it. It looks as follows:
;; QUESTION SECTION:
;_sip._udp.aimcidfilter.com. IN SRV
;; ANSWER SECTION:
_sip._udp.aimcidfilter.com. 60 IN SRV 1 100 5061 spinner.eng.rr.com.
_sip._udp.aimcidfilter.com. 60 IN SRV 2 100 5061 bart.eng.rr.com.
_sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 homer.eng.rr.com.
_sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 crunch.eng.rr.com.
3) After fetching the SRV record, the OpenSER sees that
spinner.eng.rr.com has a priority of 1 so it does a gethostbyname
which performs a DNS A record
lookup and gets a response for spinner.eng.rr.com.
4) When the OpenSER sends towards spinner.eng.rr.com, it does not get
a SIP response from it. *It is at this point I was hoping that OpenSER
would blacklist this UAS endpoint*, but for each incoming request that
is received by the OpenSER, it continues to resolve the domain in the
same manner and sends towards spinner.eng.rr.com even though there is
no SIP response.
Is what I am trying to do in accordance with how OpenSER blacklisting
is supposed to work?
I am trying to have OpenSER send towards bart.eng.rr.com when
spinner.eng.rr.com is not responding or is not reachable.
My configuration is as below. Note that I have even tried to disable
sending towards spinner.eng.rr.com by manually adding a dst_blacklist
entry - that doesn't even seem to work for me.. Am I doing something
wrong?
Here is my output from openserctl
# openserctl fifo list_blacklists
200 OK
List:: net_filter owner=13 flags=1
Rule:: flags=0
IP:: 65.185.233.55
Mask:: 255.255.255.255
Proto:: 0
Port:: 5061
List:: dns owner=17 flags=6
---------------
#
# Openser.cfg
# ----------- global configuration parameters ------------------------
debug=5 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
maxbuffer=1048576
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#reply_to_via=1
children=4
log_facility=LOG_LOCAL4
dns_use_search_list=no
dns_servers_no=2
user="sipproxy"
group="sipproxy"
disable_dns_blacklist=no
disable_dns_failover=no
dns_try_ipv6=no
dns_retr_time=2
dns_retr_no=2
dst_blacklist = net_filter:{ ( any , 65.185.233.55, 5061 , "" )} #
block towards spinner
listen=udp:65.185.232.62:5060
alias=65.185.233.104:5060
# LOAD OpenSER MODULES
mpath="/sw/lib/openser/modules/"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "acc.so"
loadmodule "mi_fifo.so"
loadmodule "xlog.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
modparam("acc", "log_level", 2)
modparam("acc", "log_flag", 1)
modparam("acc","report_cancels", 1)
modparam("acc","failed_transaction_flag", 1)
modparam("acc","log_extra", "req_uri=$rU")
modparam("tm", "fr_timer", 5)
modparam("tm","fr_inv_timer",5)
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER")
record_route();
# subsequent messages within a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
route(1);
};
if (method=="OPTIONS") {
sl_send_reply("200", "OK");
exit;
}
if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") {
xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n");
seturi("sip:feature_fs@aimcidfilter.com");
setflag(1);
route(1);
} else {
sl_send_reply("404", "Not Found");
};
}
#####################################################
# Default Message Handler
#####################################################
route[1] {
# Send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
Hello all,
Actually i have a testing setup which facilitates to give accounting
details to mysql db once call was end by any one of the uac.
But if i want to have a prepaid solution like our mobiles i dont
found any solution till now on open source
some of my friend suggested to go for B2BUA from vovida.org
does anybody using this ?
is this real that this can work for prepaid stuff?
if not just let me know otherwise i will waste my time in doing
some research on something not useful
if not what might be the other ways to get around the problem
of prepaid billing
i know we can do this using b2bua, can we go ahead with asterisk?
waiting for valuable suggestions
--
Srinivas Antarvedi
HI,
In FreeBSD 6.2, I got this error when I compiled openser with freeradius support.
../../radius.h:36:32: freeradius-client.h: No such file or directory
acc.c: In function `init_acc_rad':
acc.c:464: warning: assignment makes pointer from integer without a cast
acc.c:475: error: `DICT_ATTR' undeclared (first use in this function)
acc.c:475: error: (Each undeclared identifier is reported only once
acc.c:475: error: for each function it appears in.)
acc.c:475: error: `da' undeclared (first use in this function)
acc.c:475: error: `DICT_VALUE' undeclared (first use in this function)
acc.c:475: error: `dv' undeclared (first use in this function)
acc.c: In function `acc_rad_request':
acc.c:509: error: `VALUE_PAIR' undeclared (first use in this function)
acc.c:514: error: invalid lvalue in assignment
acc.c:555: error: `OK_RC' undeclared (first use in this function)
gmake[1]: *** [acc.o] Error 1
gmake[1]: Leaving directory `/usr/ports/net/openser/work/openser-1.2.2-tls/modules/acc'
gmake: *** [modules] Error 2
*** Error code 2
Stop in /usr/ports/net/openser.
*** Error code 1
Is there any way I can compile openser with freeradius support?
Thanks
SW