Hi all,
I need to bother about crazy client by considering "Flood" detection
technique. I can do it by using OpenSER Pike
<http://kamailio.org/docs/modules/1.2.x/pike.html> module which helps to
keep trace of all (or selected ones) incoming request's IP source and blocks
the ones that exceeded some limit.
In my case: If the number of SIP messages from a single IP address to my SIP
Proxy exceeds 200 per minute. Recommended action: Block IP for 2 hours.
I tried with the pike module but I'm little bit confused with sampling,
density, and timeout value.
Please help me with example configuration by considering my point.
Thanks,
ARIF
Hi!
I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend
is postgresql version 8.1.3.
Today I got errors in logfiles, saying:
messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]:
ERROR:avpops:dbrow2avp: dbrow contains NULL fields
The similar problem reported in:
http://lists.iptel.org/pipermail/serusers/2005-May/019681.html
with much more detailed description of error, database contents
and config samples.
Patch is trivial, and looks more like a fix to copy'n'paste error:
in mysql/val.c function str2val states:
if (!_s) {
memset(_v, 0, sizeof(db_val_t));
VAL_TYPE(_v) = _t;
VAL_NULL(_v) = 1;
return 0;
}
VAL_NULL(_v) = 0;
and the last line mentions that 'well, that's value is not NULL'.
In postgresql/db_val.c, line 182, function str2valp, the same statement is the:
if (!_s) {
DLOG("str2valp", "got a null value");
VAL_TYPE(_v) = _t;
VAL_NULL(_v) = 1;
return 0;
}
without explicit notification that this is not-NULL value.
More than, nowhere else in this function VAL_NULL(_v) not set to 0.
So, if a value _v.val contained anyting but 0 at the function start,
resulting value will be threated as NULL despite the fact, that _s is NOT NULL.
Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and
everyting will be ok.
At least for me it's ok for some hours :)
Hello Friends,
Can any one help me installing the CDR TOOL to integrate with freeradius
on freebsd? Any help will be highly appreciated.
Thanks
Mohit
Mohit C. Saxena
Sr. Network Administrator
Starcomms Plc.
www.starcomms.com <blocked::http://www.starcomms.com>
Mob: +234-702-800-0709
DISCLAIMER: The information contained in this message (including any attachments) is confidential and may be privileged. If you have received it by mistake please notify the sender by return e-mail and permanently delete this message and any attachments from your system. Any form of dissemination, use, review, distribution, printing or copying of this message in whole or in part is strictly prohibited if you are not the intended recipient of this e-mail. Please note that e-mails are susceptible to change. STARCOMMS PLC shall not be liable for the improper or incomplete transmission of the information contained in this communication nor for any delay in its receipt or damage to your system. STARCOMMS PLC does not guarantee that the integrity of this communication has been maintained or that this communication is free of viruses, interceptions or interferences. STARCOMMS PLC reserves the right to monitor all e-mail communications, whether related to the business of STARCOMMS or not, through its internal or external networks.
It would be really nice to have a feature in TM for even faster failover
that is turned off when any reply is received at all, provisional or
otherwise.
The problem with fr_timer and fr_inv_timer is that they throw on lack of
final reply. But sometimes gateways have very lengthy, complicated
asynchronous operations that must complete before they can return a
final OK. If you are doing failover, the most important thing to know
is that the gateway is there at all, which the provisional 1xx message
accomplishes.
What would be really nice is to have a timer that can be diffused as
soon as a provisional reply arrives.
I don't see how this can be accomplished with any of the existing timers.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
Hi
I have installed CDRTool and i got error on CDR search entry like this
Table 'cdrtool.radacct200810' doesn't exist
i have read on goolge about it and people tell the problem was if radacct200810 table not in radius database.
i have also run rotatetable.php script to rotate database but i got error like this
#php /var/www/CDRTool/scripts/SER/rotateTables.php
rotateTable(radacct200810,200809,radacct200809)
Error: cannot locate mysql creation file
i am using 6.6.10
kindly suggest me..
$ cat ~/satish/url.txt
http://www.linuxbug.org
_____________________________________________________________________________________________________
Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/
Hi,
I am relaying invites from Kamilio(1.2) to a third proxy, i am receivng
Server error occured in SIP messages, I am seeing these messages in log
file;
Asterisk--------->Kamilio----------------->third-proxy
route{
openser.cfg
if(method=="INVITE"){
route(4);
exit;
}
route[4] {
if(!t_relay("udp:third_proxy_1:5060") || !t_relay("udp:third_proxy_2:5060"))
{
end_media_session();
sl_reply_error();
}
}
DEBUG: RFC3261 transaction matching failed
DEBUG: t_lookup_request: no transaction found
ERROR:tm:t_forward_nonack: no branch for forwarding
ERROR:tm:w_t_relay: t_forward_nonack failed
DEBUG:sl:sl_reply_error: error text is Server error occurred (1/SL)
any idea ?
Thanks
Asim
Hi list, I am trying to solve my nat problem with openser 1.3.2 and rtpproxy 1.1, I have my openser with 2 net cards. I detail my scenario:
(192.168.10.1) LAN-eth1- Server Openser eth0-WAN (192.168.1.64)<-> NAT <-> ADSL dyndns <-> Internet <-> ADSL <-> NAT <-> UAC
All my external clients are also behind an ADSL with address private ip and my server openser, I don't have ip it public but register a domain with dyndns and it configures it in my router adsl, I have access from out to my server through dyndns.
All my external clients configure them so that they use of address proxy the dyndns domain, my external clients register but when they call to another UAC that this behind the server openser is not audio, and if a UAC that this behind the server openser calls an external client the call doesn't arrive he fails.
I have open the ports UDP 5060:5065, 10000:20000, 35000:65000
TCP: 5060
some idea of like I can solve this problem?...
best regards all list
rickygm
### LOG SIP ###
U +1.744617 192.168.10.30:5062 -> 192.168.10.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.0
Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299
Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:113@192.168.10.30:5062;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 214
v=0
o=113 8000 8000 IN IP4 192.168.10.30
s=SIP Call
c=IN IP4 192.168.10.30
t=0 0
m=audio 5004 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
#
U +0.000481 192.168.1.64:5060 -> 190.184.35.4:5063
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-de71d299
Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 102 INVITE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:113@192.168.10.30:5062;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer
Content-Length: 232
P-hint: onreply_route|force_rtp_proxy
v=0
o=113 8000 8000 IN IP4 192.168.10.30
s=SIP Call
c=IN IP4 192.168.1.64
t=0 0
m=audio 35006 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=nortpproxy:yes
#
U +0.219110 190.184.35.4:5063 -> 192.168.1.64:5060
ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-547056da
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 102 ACK
Max-Forwards: 70
Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d"
Contact: <sip:122@192.168.0.60:5063>
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0
#
U +0.000944 192.168.10.1:5060 -> 192.168.10.30:5062
ACK sip:113@192.168.10.30:5062;transport=udp SIP/2.0
Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK2fc9.6c704556.2
Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-547056da
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 102 ACK
Max-Forwards: 69
Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@gnuforever.homelinux.com",algorithm=MD5,response="ec45ed24126b924160da73b8ba10e73d"
Contact: <sip:122@190.184.35.4:5063;nat=yes>
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0
#
U +0.140447 192.168.1.64:5060 -> 190.184.35.4:5063
OPTIONS sip:190.184.35.4:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0
From: sip:pinger@192.168.1.64;tag=f7508b75
To: sip:190.184.35.4:5063
Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64
CSeq: 1 OPTIONS
Content-Length: 0
#
U +0.164881 190.184.35.4:5063 -> 192.168.1.64:5060
SIP/2.0 404 Not Found
To: sip:190.184.35.4:5063;tag=a04ea06caeb3256i3
From: sip:pinger@192.168.1.64;tag=f7508b75
Call-ID: 29b7fe71-06901b84-fa6(a)192.168.1.64
CSeq: 1 OPTIONS
Via: SIP/2.0/UDP 192.168.1.64:5060;branch=0
Server: Linksys/SPA942-5.2.8
Content-Length: 0
#
U +1.195009 192.168.10.28:5060 -> 192.168.10.1:5060
#
U +0.639899 192.168.10.1:5060 -> 192.168.10.27:5060
OPTIONS sip:192.168.10.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0
From: sip:pinger@192.168.1.64;tag=08508b75
To: sip:192.168.10.27:5060
Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1
CSeq: 1 OPTIONS
Content-Length: 0
#
U +0.001412 192.168.10.27:5060 -> 192.168.10.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=0
From: sip:pinger@192.168.1.64;tag=08508b75
To: sip:192.168.10.27:5060;tag=b1e56101b3b09b53
Call-ID: 29b7fe71-16901b84-1b6(a)192.168.10.1
CSeq: 1 OPTIONS
User-Agent: Grandstream GXV3000 1.1.3.14
Contact: <sip:120@192.168.10.27:5060>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Supported: replaces, timer, 100rel, path
Content-Length: 0
#
U +3.393457 190.184.35.4:5063 -> 192.168.1.64:5060
BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.60:5063;branch=z9hG4bK-f7539b7
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 103 BYE
Max-Forwards: 70
Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>, <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1"
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0
#
U +0.000768 192.168.10.1:5060 -> 192.168.10.30:5062
BYE sip:113@192.168.10.30:5062;transport=udp SIP/2.0
Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0
Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 103 BYE
Max-Forwards: 69
Proxy-Authorization: Digest username="122",realm="gnuforever.homelinux.com",nonce="48f260b8529fd8e9d8f0247fa92f734a317f2da5",uri="sip:113@192.168.10.30:5062",algorithm=MD5,response="4416adbfacebc76ed9cc3f002ff958a1"
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0
#
U +0.055972 192.168.10.30:5062 -> 192.168.10.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK3fc9.1cba8107.0
Via: SIP/2.0/UDP 192.168.0.60:5063;rport=5063;received=190.184.35.4;branch=z9hG4bK-f7539b7
Record-Route: <sip:192.168.10.1;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
Record-Route: <sip:192.168.1.64;r2=on;lr=on;ftag=636fc42148cbcd9ao3>
From: <sip:122@gnuforever.homelinux.com>;tag=636fc42148cbcd9ao3
To: <sip:113@gnuforever.homelinux.com>;tag=a5f269a554788978
Call-ID: 6f8c5fed-1d1d043e(a)192.168.0.60
CSeq: 103 BYE
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:113@192.168.10.30:5062;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0
Hi Castillo,I did post my config file showing how i used the Mediaproxy.
However i would like to get your opinion on how to use/configure it
properly.
cheers,
Lu.
> Message: 3
> Date: Sun, 12 Oct 2008 23:45:35 +0200
> From: I?aki Baz Castillo <ibc(a)aliax.net>
> Subject: Re: [Kamailio-Users] No audio both ways
> To: users(a)lists.kamailio.org
> Message-ID: <200810122345.35371.ibc(a)aliax.net>
> Content-Type: text/plain; charset="utf-8"
>
> El Domingo, 12 de Octubre de 2008, luzango mfupe escribi?:
> > I solved the problem by disabling use_mediaproxy() call and did put
> nat=yes
> > in the Asterisk side.
>
> This means that you didn't use MediaProxy correctly. If it's properly
> configured and the proxy logic script calls use_mediaproxy() when needed,
> then you don't need to enable the Comedia mode in Asterisk (nat=yes).
>
> --
> I?aki Baz Castillo
>
>
>
> ------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>
>
> End of Users Digest, Vol 41, Issue 33
> *************************************
>
--
Luzango Mfupe
TUUNE MOBILE
Tel:0128440528/0123825710
Tshwane-RSA
"...Ships are safe in harbor, but they were never meant to stay
there......."
Hi mates,
I solved the problem by disabling use_mediaproxy() call and did put
nat=yesin the Asterisk side.
Many thanks to Daniel and David.
Cheers,
Lu.
--
Luzango Mfupe
TUUNE MOBILE
Tel:0128440528/0123825710
Tshwane-RSA
"...Ships are safe in harbor, but they were never meant to stay
there......."
by accident i sent this message to devel list when i intended to send it
to users list.
i'm thinking to speed up lcr load_gws() implementation by storing prefixes
into hash table. it would allow load_gws() to execute (if necessary) in
constant time independent of the number of prefixes. also, i'm planing
to make the size of hash table controllable by module parameter.
the side effect of this is that prefix_mode=1, where prefixes are
regular expressions, would have to be dropped, because hashing on
regular expressions makes no sense.
something close to prefix_mode=1 could be achieved using dialplan module
to store the regular expressions and return as attribute a gateway
grp_id. that could then be used by new load_gws_from_group(grp_id) to
load gws in the given group.
any comments on this? is someone depending on prefix_mode=1?
by the way, before starting this project, i took a look at opensips
drouting module, but unfortunately it does not cover many features of
lcr module that i use in my configurations and thus cannot be used to
replace lcr module.
-- juha