[OpenSIPS-Users] Getting Error when using NATHELPER module

2010-05-03 Thread Ahmed Munir
Hi,

I'm getting error when I configured Nathelper on OpenSIPs, the errors are
listed down below;

May  3 05:44:43 newtest /usr/local/sbin/opensips[2999]:
ERROR:nathelper:select_rtpp_node: script error -no valid set selected
May  3 05:44:43 newtest /usr/local/sbin/opensips[2999]:
ERROR:nathelper:force_rtp_proxy_body: no available proxies
May  3 05:45:04 newtest /usr/local/sbin/opensips[2998]:
ERROR:nathelper:force_rtp_proxy: Unable to parse body

and my configuration of OpenSIPs is listed below;

loadmodule "dispatcher.so"
loadmodule "avpops.so"
loadmodule "permissions.so"
loadmodule "aaa_radius.so"
loadmodule "auth_aaa.so"
#loadmodule "auth_diameter.so"
loadmodule "nathelper.so"


#Settings For
Radius-
#modparam("auth_diameter", "diameter_client_host", "localhost")
modparam("aaa_radius",
"radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "aaa_url",
"radius:/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "aaa_flag", 2)
modparam("acc", "aaa_missed_flag", 3)
modparam("acc", "aaa_extra","User-Name=$Au; \
Calling-Station-Id=$from; \
Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ruri; \
Sip-RPid=$avp(s:rpid); \
Source-IP=$si; \
Source-Port=$sp; \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
X-RTP-Stat=$hdr(X-RTP-Stat); \
Contact=$hdr(contact); \
Event=$hdr(event); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
ENUM-TLD=$avp(s:enum_tld)")

modparam("auth_aaa","aaa_url","radius:/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("auth", "rpid_prefix", ";screen=yes;privacy=off")
#modparam("auth", "rpid_suffix", "@203.215.179.54>;screen=yes;privacy=off")
modparam("auth", "rpid_avp", "$avp(s:rpid)")
#modparam("uri","service_type",10)


# - setting module-specific parameters ---

modparam("dispatcher", "db_url", "mysql://opensips:opensip...@localhost
/opensips")
modparam("permissions", "db_url", "mysql://opensips:opensip...@localhost
/opensips")

#- setting NAT module parameters -

modparam("nathelper","ping_nated_only",1)
modparam("nathelper", "natping_interval", 30)
modparam("nathelper","natping_processes",1)
#modparam("nathelper","rtpproxy_sock","udp:127.0.0.1:7890")
modparam("nathelper","received_avp","$avp(i:42)")
#modparam("nathelper", "sipping_bflag", 7)
modparam("usrloc", "nat_bflag", 6)

route{

if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

#NAT detection
log("# Go to Route 3 for NAT
Detection #");
route(3);

if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction
fails
} else if (is_method("INVITE")) {
record_route();
}

route(1);
} else {

if ( is_method("ACK") ) {
if ( t_check_trans() ) {
t_relay();
exit;
} else {
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}

#initial requests

# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}

t_check_trans();

if (loose_route()) {
xlog("L_ERR",
"Attempt to route with preloaded Route's
[$fu/$tu/$ru/$ci]");
if (!is_method("ACK"))
sl_send_reply("403","Preload Route denied");
exit;
}

# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();

$avp(s:checksrc) = check_source_address("0");

log("##

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir
Hi,

Thanks for replying. Can you please check my configuration of OpenSIPs what
I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.

Please point out in which section do I required to add force_rtp_proxy(),
because I already configured Nat on it. kindly advise me soon.

On Fri, Apr 30, 2010 at 11:35 AM,  wrote:

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> Today's Topics:
>
>   1. Re: NAT Problem using Nat helper (Laszlo)
>
>
> --
>
> Message: 1
> Date: Fri, 30 Apr 2010 08:35:00 +0200
> From: Laszlo 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID:
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Ahmed,
>
> As you can see, the other party gets local ip in SDP
>
> c=IN IP4 192.168.0.168.
>
> You can try to play with flags:
> http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028
>
> -Laszlo
>
>
>
> 2010/4/30 Ahmed Munir 
>
> >
> >
> > Hi.
> >
> > Thanks for your reply, the traces are metioned below;
> >
> > U 203.215.176.22:55134 -> 11.22.33.44:5060
> > .
> > .
> > ..
> >
> > U 81.201.82.45:5060 -> 11.22.33.44:5060
> > INVITE sip:1234...@11.22.33.44  <
> sip%3a1234...@11.22.33.44 > SIP/2.0.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@11.22.33.44 >>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Max-Forwards: 69.
> > Content-Type: application/sdp.
> > Contact: .
> > User-Agent: Vox Callcontrol.
> > Content-Length: 210.
> > .
> > v=0.
> > o=root 13293 13293 IN IP4 81.201.82.146.
> > s=session.
> > c=IN IP4 81.201.82.146.
> > t=0 0.
> > m=audio 11458 RTP/AVP 8 0.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> >
> > U 11.22.33.44:5060 -> 81.201.82.45:5060
> > SIP/2.0 100 Giving a try.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@11.22.33.44 >>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
> > Server: OpenSIPS (1.6.1-notls (i386/linux)).
> > Content-Length: 0.
> > .
> >
> >
> > U 11.22.33.44:5060 -> 203.215.176.22:55134
> > INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26 SIP/2.0.
> > Record-Route: .
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > From: "4572727220" 
> >
> > >;tag=43772.
> > To:  <
> sip%3a1234...@11.22.33.44 >>.
> > Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> >
> ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Max-Forwards: 68.
> > Content-Type: application/sdp.
> > Contact: .
> > User-Agent: Vox Callcontrol.
> > Content-Length: 210.
> > P-hint: usrloc applied.
> > .
> > v=0.
> > o=root 13293 13293 IN IP4 81.201.82.146.
> > s=session.
> > c=IN IP4 81.201.82.146.
> > t=0 0.
> > m=audio 11458 RTP/AVP 8 0.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> >
> > U 203.215.176.22:55134 -> 11.22.33.44:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285.0.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> >
> ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Record-Route: .
> > Contact: .
> > To:  <
> sip%3a1234...@11.22.33.44 >>;tag=611cee1e.
> > From: "4572727220"
> >
> > >;tag=43772.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > User-Agent: X-Lite release 1104o stamp 56125.
> > Content-Length: 0.
> > .
> >
> >
> > U 11.22.33.44:5060 -> 81.201.82.45:5060
> > SIP/2.0 180 Ringing.
> > Via: SIP/2.0/UDP 81.201.82.45:5060
> >
> ;rport=5060;received=81.201.82.45;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Record-Route: .
> > Contact:  > ;rinstance=25bfe05618433c26;nat=yes>.
> > To:  <
> sip%3a1234...@11.22.33.44 >>;tag=611cee1e.
> > From: "4572727220"
> >
> > >;tag=43772.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45.
> > CSeq: 102 INVITE.
> > User-Agent: X-Lite release 1104o stamp 56125.
> > Content-Length: 0.
> > .
> >
> >
> > U 203.215.176.22:55134 -> 11.22.33.44:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP 11.22.33.44;branch=z9hG4bK1c7c.78b70285

Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RFC] OpenSIPS 2.0 - changes on the SIP messages

2010-05-03 Thread Emmanuel BUU
Hello,

First of all, I want to stress that the whole concept of message 
scripting is a fabulous asset. It enables us
to adapt to so many situations without recompling C code. I have the 
feeling that the whole concept has emerged from a rather practical
and performance oriented approach (which is good). We should maybe look 
at what we idealy would like:

We want to be able to modify the message and see the modification "in 
real time". For some values,
we also want either to undo some changes (mainly the module developpers) 
or access the original values
the scripts writers.

So IMHO there should be some place where the original message stand, 
preparsed by the core. The modifications should be applied on a 
preparsed copy
and the module developper should be able to access both but modify only 
the one that is intented to handle the modification. At the end of the 
processing
the modified message should be translated from a parsed form to a text 
buffer.

This process might be slower than the original lump model that is highly 
optimized but it may lead to a more sold ground.

I agree with the remark about branch isolation.

Emmauel

Bogdan-Andrei Iancu a écrit :
> Hi,
>
> sorry for crossposting, but I thing this topic is interesting both for 
> users (as experience) and for developers (as solution).
>
> So, right now the code for 2.0 got to a stage were we need to "apply" 
> changes on the received messages (like adding VIA headers, changing 
> headers, etc). And before starting the work on this area, I would like 
> to get some opinions on the available options:
>
> 1) keep the current mechanism for changing the SIP messages by using 
> lumps that are applied only when the message is sent out :
>Advantages: code is there and works; the mechanism is very fast ; 
> no need to re-parse the message after the parsing
>Disadvantages : from script or code, you are not able to see the 
> previous changes you did on the message, like:
>  if you remove a hdr, it will be marked to be removed, 
> but you will still see it there after the removed
>  if you add a new hdr, you will not be able to see it 
> (it will be actually added only when the message will be sent out)
>  if you replace the contact hdr (like after 
> fix_nated_contact() ) will not see the new contact, but the old one
>  trying to apply 2 changes in the same area (like 
> changing twice the contact) will result in bogus result.
>
> 2) find a new approach that will allow to push in realtime the 
> changes on messages
>Advantages: the change will take affect instantly, so all the 
> disadvantages from 1) (as user experience in operating opensips) will 
> disappear .
>Disadvantages : the code will probably be slower (the changes 
> part), but will speed up other parts of the code (where you need to 
> manually identify the prev changes);
>  changed parts of the SIP message will require re-parsing 
> (so the code will have to check the state of a header (if parsed or not) 
> before each usage)
>  you will not be able to undo changes on the SIP messages in 
> an easy way (for example during the parallel and serial forking on 
> per-branch changes)
>
> So this is the dilemma: if the current mechanism (with applying changes 
> at the end) such a bad user experience (scripting) in order to try for 
> 2.0 a different approach ?
>
> I would like to hear some opinions from both people writing scripts and 
> people writing code.
>
> Thanks and regards,
> Bogdan
>
>   


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[OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.conf doesn't exist

2010-05-03 Thread vishu
Directory  /etc/opensips/radius is generated at Opensips installation time
or Freeradius installation time ??

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[OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.confdoesn't exist

2010-05-03 Thread vishu
So should I copy clients.conf from /etc/freeradius/clients.conf to
/etc/opensips/radisus/client.conf  or can define the path
/etc/freeradius/clients.conf in opensips.conf , does it matter ? 

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[OpenSIPS-Users] install cdrtools

2010-05-03 Thread kenny good
Dear
   who can give me file of cdrtools installation
   I install opensips with mysql & callcontrol
   & install callcontrol app
   But I cann't install CDRTools 

   apt-get install cdrtool
   tell me need geoip-database,OK I installed ,and do it again
   tell me need php5-geoip,OK installingbut this step remove the 
geoip-database

   My god? How to do?


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Re: [OpenSIPS-Users] OpenSIPS defunct processes

2010-05-03 Thread Bogdan-Andrei Iancu
Hi David,

Based on the "ps" output, it seams that the zombies processes were 
forked by opensips worker processes - this does not happen only when 
using the exec module (which you do not have) - the only alternative is 
that the perl scripts you are using are doing the fork (maybe some perl 
function?) and do not properly terminate the extra procs...

Regards,
Bogdan

David Cunningham wrote:
> Hello,
>
> Certainly, here they are from opensips.cfg and I've included the
> modparam in case they help:
>
> loadmodule "db_mysql.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> loadmodule "usrloc.so"
> loadmodule "auth.so"
> loadmodule "auth_db.so"
> loadmodule "maxfwd.so"
> loadmodule "mi_fifo.so"
> loadmodule "nathelper.so"
> loadmodule "perl.so"
> loadmodule "registrar.so"
> loadmodule "rr.so"
> loadmodule "textops.so"
> loadmodule "uri.so"
>
> modparam( "auth", "nonce_expire", 30 )
> modparam( "auth_db|domain|uri_db|usrloc", "db_url", "mysql://foo" )
> modparam( "auth_db", "calculate_ha1", yes )
> modparam( "auth_db", "password_column", "secret" )
> modparam( "auth_db", "use_domain", 0 )
> modparam( "auth_db", "user_column", "name" )
> modparam( "mi_fifo", "fifo_name", "/tmp/opensips_fifo" )
> modparam( "nathelper", "natping_interval", 240 )
> modparam( "nathelper", "ping_nated_only", 1 )
> modparam( "nathelper", "sipping_bflag", 1 )
> modparam( "nathelper", "sipping_from", "sip:keepal...@foo" )
> modparam( "nathelper|registrar", "received_avp", "$avp(i:42)" )
> modparam( "perl", "filename", "/path/to/OpenSIPS.pm" )
> modparam( "perl", "modpath", "/path/to/perllib" )
> modparam( "registrar", "append_branches", 1 )
> modparam( "rr", "enable_full_lr", 1 )
> modparam( "usrloc", "db_mode", 2 )
> modparam( "usrloc", "desc_time_order", 1 )
> modparam( "usrloc", "nat_bflag", 1 )
> modparam( "usrloc", "timer_interval", 5 )
>
>
> Thank you!
>
> On Wed, Apr 28, 2010 at 4:53 PM, Bogdan-Andrei Iancu
>  wrote:
>   
>> Hi David,
>>
>> by chance, using the "exec" module  ?
>>
>> Or, can you list the modules you are using ?
>>
>> Regards,
>> Bogdan
>>
>> David Cunningham wrote:
>> 
>>> Hello,
>>>
>>> Thank you for the reply. I checked the parent of the zombie processes,
>>> and they seem to be "SIP receiver" processes as per the following "ps
>>> -ef" extract and "opensipsctl fifo ps" information.
>>> We're not running the "respawn" patch.
>>>
>>> Any more advice very welcome, thanks again!
>>>
>>>
>>> user  5830 1  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5832  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5833  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5834  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5835  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5836  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5837  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5838  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5839  5830  0 06:38 ?00:00:16 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5840  5830  0 06:38 ?00:00:01 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5841  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5842  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5843  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5844  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5845  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5846  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5847  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5848  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5849  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5850  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  5851  5830  0 06:38 ?00:00:00 /sbin/opensips -m 256 -P
>>> /var/run/user/opensips.pid
>>> user  7260  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  7261  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  7262  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  7263  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  7264  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  7265  5833  0 08:30 ?00:00:00 [opensips] 
>>> user  9770  5835  0 08:3

Re: [OpenSIPS-Users] Command "opensipsctl dialplan reload" randomly hangs

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Dan,

The vital information is the backtrace (with gdb) of the FIFO process 
while blocked = as there is not CPU usage, I say the procs stuck  in a 
IO op. BTW, try to run the test with debug=6 (if possible).

If you need help with the gdb syntax, let me know.

Regards,
Bogdan

DanB wrote:
> Hey Bogdan,
>
> Hereby some more tests (I should mention that I cannot block on demand
> the fifo, it simply randomly blocks, so I must be lucky when I am
> trying to reproduce it).
>
> Based on ps the cpu and memory consuption do not increase during the
> hang (CPU:0.0  Memory:1.0).
>
> Ta,
> DanB
>
> Log of the actions:
> ---
>
> Before reload:
>
> sip1:~# opensipsctl ps|grep FIFO
>
> Process::  ID=13 PID=26368 Type=MI FIFO
>
> sip1:~# ps uww -p 26368
>
> USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
>
> opensips 26368  0.0  1.0 1147236 10788 ?   S07:30   0:00
> /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g
> opensips
>
> sip1:~#
>
>
>
> During hanging reload:
>
> sip1:~# ps uww -p 26368
>
> USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
>
> opensips 26368  0.0  1.0 1147644 11040 ?   R07:30   0:00
> /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g
> opensips
>
>
> After reload interrupted with crtl+c and /etc/init.d/opensips restart
>
>
> sip1:~# ps uww -p 26368
>
> USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
>
> sip1:~# opensipsctl ps|grep FIFO
>
> Process::  ID=13 PID=26716 Type=MI FIFO
>
> sip1:~# ps uww -p 26716
>
> USER   PID %CPU %MEMVSZ   RSS TTY  STAT START   TIME COMMAND
>
> opensips 26716  0.0  0.1 1147164 1988 ?S14:59   0:00
> /usr/sbin/opensips -P /var/run/opensips/opensips.pid -m 1024 -u opensips -g
> opensips
>
> On Mon, Apr 26, 2010 at 12:25 PM, Bogdan-Andrei Iancu
>  wrote:
>   
>> Hi Dan,
>>
>> Your descriptions point to a blocked fifo process. Blocking maybe
>> because of some internal locking (you see 99% cpu usage) or some I/O
>> (normal cpu usage).
>>
>> So, do the followings:
>>
>> 1) do "opensipsctl fifo ps" to see the PID of the fifo process
>> 2) make fifo to block
>> 3) check if the fifo process (by pid) is there - if yes, see how much
>> cpu it uses and try to attache with gdb to it to get a backtrace.
>>
>> Regards,
>> Bogdan
>>
>> DanB wrote:
>> 
>>> Hey Bogdan,
>>>
>>> Thanks for coming back so fast.
>>>
>>> There was no error reported neither on console nor in the syslog
>>> (debug 7). I will need to check for dead process since all I could
>>> spot was no reply back and console hanging, and be able to stop it
>>> only with Ctrl+C. After Ctrl+C could not get the any other opensipsctl
>>> commands to work.
>>> Will need to wait few days more to know about dead process.
>>>
>>> DanB
>>>
>>> On Thu, Apr 22, 2010 at 6:55 PM, Bogdan-Andrei Iancu
>>>  wrote:
>>>
>>>   
 Hi Dan,

 Did you notice any error from the fifo process during the reload ? it
 may be something related to locking (during reload) of the table -> this
 may affect all the other processes.

 So, any errors? any dead processes (like fifo one) ?

 Regards,
 Bogdan

 DanB wrote:

 
> Hey Guys,
>
> I have noticed the "opensipsctl dialplan reload" command randomly
> hanging, sometimes even the server itself becoming non responsible,
> other times reloading the dialplan into memory but not reporting
> anything on console, the last one becoming unusable until server
> restart. This happened in the past as well but with the traffic
> increase, it becomes more and more annoying. I suspect the same bug
> which was present in the past with fifo hanging.
> I should mention that I got about 2000 records in the dialplan table,
> so I would say not that much loaded.
>
> The version I am running:
> sip1:/home/employee/dan# opensips -V
> version: opensips 1.6.1-notls (x86_64/linux)
> flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST,
> 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
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> svnrevision: 2:6509M
> @(#) $Id: main.c 6169 2009-09-22 12:48:37Z bogdan_iancu $
> main.c compiled on 14:43:30 Jan 11 2010 with gcc 4.3.2
>
>
> All I could find in the logs was database reconnection in the case of
> hang but no error or something else reported (running debug 7).
> Bellow some of the log:
> Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]:
> DBG:mi_fifo:mi_fifo_server: entered consume
> Apr 22 14:27:19 sip1 /usr/sbin/opensips[16299]:
> DBG:mi_fifo:mi_fifo_server:  done consume
> Apr 22 14:27:19 sip1 /

Re: [OpenSIPS-Users] Opensips 1.6.2 w/ Control Panel 4.0 - Problem w/ CDRViews and Dialog

2010-05-03 Thread Bogdan-Andrei Iancu
And what is your error in the dialog ad cdrviewer modules ?? what is not 
working ? any error message ?

Regards,
Bogdan

Erick Chinchilla Berrocal wrote:
>
>  
>
>  
>
> *From:* Erick Chinchilla Berrocal [mailto:er...@netcrc.net]
> *Sent:* Wednesday, April 28, 2010 11:08 AM
> *To:* 'OpenSIPS users mailling list'
> *Subject:* RE: [OpenSIPS-Users] Opensips 1.6.2 w/ Control Panel 4.0 - 
> Problem w/ CDRViews and Dialog
>
>  
>
> Alex this is my current setup
>
> -  /var/www/opensips-cp/config/db.inc.php
>
> //database driver mysql or pgsql
>
> $config->db_driver = "mysql"; 
>
>  
>
>  //database host
>
>  $config->db_host = "localhost";
>
>  
>
>  //database port - leave empty for default
>
>  $config->db_port = "";
>
>  
>
>  //database connection user
>
> $config->db_user = "root";
>
>  
>
>  //database connection password
>
>  $config->db_pass = "passwd root";
>
>  
>
> //database name
>
> $config->db_name = "opensips";
>
>  
>
>  if (!empty($config->db_port) ) $config->db_host = $config->db_host . 
> ":" . $config->db_port;
>
>  
>
> ?>
>
>  
>
> -  /var/www/opensips-cp/config/boxes.global.inc.php
>
> /* DEFINITION OF BOXES (servers) 
> */
>
> // each server is a box
>
>  
>
> $box_id=0;
>
>  
>
> // mi host:port pair || fifo_file
>
> $boxes[$box_id]['mi']['conn']="/tmp/opensips_fifo";
>
>  
>
> // monit host:port
>
> $boxes[$box_id]['monit']['conn']="127.0.0.1:2812";
>
> $boxes[$box_id]['monit']['user']="admin";
>
> $boxes[$box_id]['monit']['pass']="monit";
>
> $boxes[$box_id]['monit']['has_ssl']=0;
>
>  
>
>  
>
> // description (appears in mi , monit )
>
> $boxes[$box_id]['desc']="Primary SIP server";
>
>  
>
>  
>
> $boxes[$box_id]['assoc_id']=1;
>
>  
>
> // enable local smonitor charts on this box : 0=disabled 1=enabled
>
> // (cron)
>
> $boxes[$box_id]['smonitor']['charts']=1;
>
>  
>
> -  Opensips.cfg
>
> -  # - mi_fifo params -
>
> -  modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
>
> -  modparam("mi_fifo", "fifo_mode", 0666)
>
>  
>
> Thanks
>
> Erick Ch.
>
>  
>
> *From:* users-boun...@lists.opensips.org 
> [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Alex Ionescu
> *Sent:* Wednesday, April 28, 2010 3:07 AM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] Opensips 1.6.2 w/ Control Panel 4.0 - 
> Problem w/ CDRViews and Dialog
>
>  
>
> Hi Erick,
>
> The global db.inc.php (the one you have in 
> /var/www/opensips-cp/config/) is used by all the modules as long as 
> they don't find something more specific defined into their config 
> files (like, for example, if you want a different db setup for module 
> *domains* you can edit 
> */var/www/opensips-cp/config/tools/system/domains/db.inc.php* - and 
> this will override the global db setup ).
>
> Dialog takes the call information using a MI command. So you must have 
> OpenSIPS Control Panel and OpenSIPS configured properly (you must 
> choose between xmlrpc and fifo) - see the OpenSIPS config file (to 
> enable mi_xmlrpc module or the mi_fifo) and also check the 
> config/boxes.global.inc.php to have the correct MI connection 
> parameters set up.
>
> Regards,
> Alex
>
>
>
> -- 
> Alex Ionescu
> www.voice-system.ro  
>
>
>
>
>
> __ Information from ESET NOD32 Antivirus, version of virus 
> signature database 3997 (20090409) __
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: [OpenSIPS-Users] install cdrtools

2010-05-03 Thread Adrian Georgescu
This works fine on Debian unstable. Are you using some other OS? May I  
ask which one?


Adrian



On Apr 29, 2010, at 9:47 AM, kenny good wrote:


Dear
   who can give me file of cdrtools installation
   I install opensips with mysql & callcontrol
   & install callcontrol app
   But I cann't install CDRTools

   apt-get install cdrtool
   tell me need geoip-database,OK I installed ,and do it again
   tell me need php5-geoip,OK installingbut this step remove  
the geoip-database


   My god? How to do?


Kenny
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Re: [OpenSIPS-Users] Are there in OpenSIPS modules like AGI on asterisk ?

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Sam,

You can use the avpops module to run custom queries from the opensips 
script. Also, exec module allows you to run external applications 
(scripts, whatever) from the opensips script.

Regards,
Bogdan

samoh wrote:
> Hi Bogdan,
>
> I really want to have a possibility to run a script on  OpenSIPS or external
> of OpenSIPS, I want to run a script in python or other language as a way to
> query my database to configure the call by calling (that is just an example)
> not to be dependent on any module.
> I am doing my internship in a company that has its own script about asterisk
> and architecture of the database and my target is to developpe an IP Centrex
> with Opensips in order to don't touch to the present architecture (Database)
> and my solution must be independent of mudules so that allows any other
> developpement on this IP Centrex.
>
> I really don't know how to do, it's some days I'm looking but no idea yet.
>
> is there a function which allows to end a call after N minut ?
>
> Thanks.
>
> Best regards.
> Sam.
>   


-- 
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www.voice-system.ro


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Re: [OpenSIPS-Users] OpenSIPS-Users] takes ages to register

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Andy,

The traces shows that opensips replies in notime to all the REGISTER 
requests - I see a 12 secs delay between the second 401 reply (from 
opensips) and REGISTER request (from UAC) with credentials -> so the 
delay comes from UAC. At least this is what the traces shows.

Regards,
Bogdan

Andy Thomas wrote:
> Heres some traces-
>
> This is my first softphone, takes 3 attempts to register, and a total of 24
> seconds to complete:
>
>
> U 2010/04/28 20:45:25.685389 79.121.180.192:38670 -> 92.63.137.209:5060
> REGISTER sip:voipexpress.co.uk SIP/2.0.
> Via: SIP/2.0/UDP 192.168.10.13:5070;rport;branch=z9hG4bK04640.
> Max-Forwards: 20.
> To: .
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 2 REGISTER.
> Contact: ;expires=3600;q=0.90.
> User-Agent: NCH Software Express Talk 4.03.
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 20:45:25.687274 92.63.137.209:5060 -> 79.121.180.192:38670
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 192.168.10.13:5070;rport=38670;branch=z9hG4bK04640;received=79.121.180.192.
> To:
> ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.a93d.
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 2 REGISTER.
> WWW-Authenticate: Digest realm="voipexpress.co.uk",
> nonce="4bd89073000b75de71bff18e6f3b1b75ea9d026629f1".
> Server: OpenSIPS (1.6.2-notls (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 20:45:37.683605 79.121.180.192:38670 -> 92.63.137.209:5060
> REGISTER sip:voipexpress.co.uk SIP/2.0.
> Via: SIP/2.0/UDP 192.168.10.13:5070;rport;branch=z9hG4bK14640.
> Max-Forwards: 20.
> To: .
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 3 REGISTER.
> Contact: ;expires=3600;q=0.90.
> User-Agent: NCH Software Express Talk 4.03.
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 20:45:37.684962 92.63.137.209:5060 -> 79.121.180.192:38670
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 192.168.10.13:5070;rport=38670;branch=z9hG4bK14640;received=79.121.180.192.
> To:
> ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ed8d.
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 3 REGISTER.
> WWW-Authenticate: Digest realm="voipexpress.co.uk",
> nonce="4bd8907f000c3e962a906408586f31fdad8586892592".
> Server: OpenSIPS (1.6.2-notls (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 20:45:49.683648 79.121.180.192:38670 -> 92.63.137.209:5060
> REGISTER sip:voipexpress.co.uk SIP/2.0.
> Via: SIP/2.0/UDP 192.168.10.13:5070;rport;branch=z9hG4bK24640.
> Max-Forwards: 20.
> To: .
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 4 REGISTER.
> Contact: ;expires=3600;q=0.90.
> Authorization: Digest
> username="testuser1",realm="voipexpress.co.uk",nonce="4bd89073000b75de71
> bff18e6f3b1b75ea9d026629f1",uri="sip:voipexpress.co.uk",response="d2dbbfb5e7
> 36286471256c963873be96",opaque="",algorithm=MD5.
> User-Agent: NCH Software Express Talk 4.03.
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 20:45:49.687049 92.63.137.209:5060 -> 79.121.180.192:38670
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 192.168.10.13:5070;rport=38670;branch=z9hG4bK24640;received=79.121.180.192.
> To:
> ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.215c.
> From: ;tag=2359.
> Call-ID: 1272483910-4640-andy-h...@192.168.10.13.
> CSeq: 4 REGISTER.
> Contact: ;q=0.9;expires=321,
> ;q=0.9;expires=3600.
> Server: OpenSIPS (1.6.2-notls (i386/linux)).
> Content-Length: 0.
> ..
>
>
>
> Here's one using X-lite (takes 30 seconds to register):
>
> U 2010/04/28 21:42:29.777132 79.121.180.192:38672 -> 92.63.137.209:5060
> REGISTER sip:voipexpress.co.uk SIP/2.0.
> Via: SIP/2.0/UDP
> 192.168.10.13:5072;branch=z9hG4bK-d8754z-664c4957b5641f55-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: .
> To: "testuser2".
> From: "testuser2";tag=9d01cb20.
> Call-ID: NTBlZmM2MWM0NGUxZDhiNWExN2E3MzhjOWZkMTU1ZmI..
> CSeq: 1 REGISTER.
> Expires: 3600.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
> INFO.
> User-Agent: X-Lite release 1104o stamp 56125.
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 21:42:29.778904 92.63.137.209:5060 -> 79.121.180.192:38672
> SIP/2.0 401 Unauthorized.
> Via: SIP/2.0/UDP
> 192.168.10.13:5072;branch=z9hG4bK-d8754z-664c4957b5641f55-1---d8754z-;rport=
> 38672;received=79.121.180.192.
> To:
> "testuser2";tag=c97b4d1cb1f3d0da549e06a8d48
> 2ef63.8145.
> From: "testuser2";tag=9d01cb20.
> Call-ID: NTBlZmM2MWM0NGUxZDhiNWExN2E3MzhjOWZkMTU1ZmI..
> CSeq: 1 REGISTER.
> WWW-Authenticate: Digest realm="voipexpress.co.uk",
> nonce="4bd89dd3004e370b6dcc60ce56bc6659bd99bd333eca".
> Server: OpenSIPS (1.6.2-notls (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 2010/04/28 21:42:59.771090 79.121.180.192:38672 -> 92.63.137.209:5060
> REGISTER sip:voipexpress.co.uk SIP/2.0.
> Via: SIP/2.0/UDP
> 192.168.10.13:5072;branch=z9hG4bK-d8754z-f827ad1654460f24-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: .
> To: "testuser2".
> From: "testuser2";tag=9d01cb20.
> C

Re: [OpenSIPS-Users] UPDATE message and sdp

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Daniel,

whatever function you are using for triggering rtpproxy, check that the 
function is called and if its execution is successful (check return code 
in the script and the errors in log). if ok, you should see in the SDP 
the rtpproxy indication and the m and c lines changed.  Right now, your 
UPDATE does not show any of these.

Regards,
Bogdan

Daniel Goepp wrote:
> I am using rtpproxy_offer and rtpproxy_answer as the doc indicated 
> that force_rtp_proxy was depreciated, however I will try it in this 
> situation to see if it works.
>
> -dg
>
>
> On Wed, Apr 28, 2010 at 2:49 AM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Daniel,
>
> Are you doing force_rtp_proxy for the UPDATE request?
>
> Regards,
> Bogdan
>
> Daniel Goepp wrote:
> > And just after replying regarding my success (I have been
> running this
> > for many months now without problems), I do have a very specific
> issue
> > related to UPDATE messages.
> >
> > I have added some logging and for the following call, I get:
> >
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21903]:
> > Request UPDATE: sip:2021@:5069 -
> > sip:6502734...@192.168.1.110:5060;transport=tcp
> > Apr 27 14:45:49 ip-10-250-14-133
> /usr/local/sbin/opensips[21903]: Route 1
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21903]:
> > Setting rtpproxy_offer - Route 1
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21903]:
> > Fixing Contact - Route 1
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21903]:
> > Relay message
> > Apr 27 14:45:49 ip-10-250-14-133
> /usr/local/sbin/opensips[21903]: new
> > branch route 1 at sip:2021@:5069  -
> > sip:6502734101@:33774;transport=tcp
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21896]:
> > incoming reply route 1 -  - sip:2...@192.168.1.101:5069
> 
> > 
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21896]:
> > Found a response from a private address! -
> sip:2...@192.168.1.101:5069 
> > 
> > Apr 27 14:45:49 ip-10-250-14-133 /usr/local/sbin/opensips[21896]:
> > Setting rtpproxy_answer - Reply 1
> > Apr 27 14:45:49 ip-10-250-14-133 rtpproxy[20092]:
> INFO:handle_command:
> > lookup on ports 30882/29328, session timer restarted
> > Apr 27 14:45:49 ip-10-250-14-133 rtpproxy[20092]:
> INFO:handle_command:
> > lookup on ports 31198/28836, session timer restarted
> >
> > Which to me looks like it is identifying that it needs to fix
> the SDP,
> > but then in the outbound UPDATE the connection IP is still the
> private
> > address.  See trace below.  The signaling seems okay otherwise, and
> > the experience that I get is that the endpoint being called to
> (2021)
> > can no longer see the calling party (we are testing video).  The 200
> > OK coming back does not have this problem, it's connection IPs
> in the
> > SDP are rewritten fine, making me think perhaps it's something
> related
> > to just UPDATE message, but I don't know enough about the inner
> > workings of OpenSIPS.  Thoughts?
> >
> > Thanks
> >
> > -dg
> >
> > 
> > 2010-04-27 14:45:49
> > tcp::33774 -> tcp::5060
> >
> > UPDATE sip:2021@:5069 SIP/2.0
> > Via: SIP/2.0/TCP
> >
> 192.168.1.110:5060;branch=z9hG4bKc5744895b5e0bbaaff08043324b99dc5.1;rport
> > Call-ID: 4fdbf3351f217...@192.168.1.110
> 
> >  >
> > CSeq: 104 UPDATE
> > Contact: 
> > From:  
> >  >>;tag=7ad977f50f0f9d94
> > To: "Daniel Goepp"  
> >  >>;tag=DC151CA5-80A23EC4
> > Max-Forwards: 70
> > Route: ;lr;transport=tcp;transport=tcp>
> > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
> > User-Agent: TANDBERG/257 (TE2.2.0.213935Beta5)
> > Proxy-Authorization: Digest nonce="*", realm="mydomain.com
> 
> > ", username="6502734101",
> uri="sip:mydomain.com 
> > ", response="***", algorithm=MD5
> > Supported: replaces,100rel,timer,gruu,path,outbound
> > Session-Expires: 500;refresher=uac
> > Min-SE: 90
> > Content-Type: application/sdp
> > Content-Length: 455
> >
> > v=0
> > o=tandberg 17 2 IN IP4 192.

Re: [OpenSIPS-Users] Are there in OpenSIPS modules like AGI on asterisk ?

2010-05-03 Thread samoh

Hi Bogdan, 

Thank you very much for your responses.
I will tray to use exec module ;).

Best regards.
Sam.
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Re: [OpenSIPS-Users] tm - module out of memory

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Julien,

try to monitor the memory consumption, as it may be a memory overload 
(too many transaction in mem) due routing problems. See
  http://www.opensips.org/Resources/DocsTsMem

So, check the available mem (use "opensipsctl fifo get_statistics all" 
), run the test until you get the out of mem error -> check how many 
transactions are in mem ; after that stop the traffic and see if the mem 
value goes back.

maybe printing a mem dump to see if there is a leak (in care of error) 
will be helpful.

Regards,
Bogdan

Julien Chavanton wrote:
>  
> I have to diagnostic this system further, as before facing 
> a memory shortage, there is a huge bunch of
> ERROR:tm:t_check: reply cannot be parsed
>  
>  
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH ROUTE**
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY ROUTE - 
> td[10.1.16.50]**
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]: 
> ERROR:tm:t_check: reply cannot be parsed
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST ROUTED ***
>
> 
> *From:* users-boun...@lists.opensips.org on behalf of Julien Chavanton
> *Sent:* Thu 29/04/2010 8:27 PM
> *To:* Users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] tm - module out of memory
>
> tm module went out of memory, we did face other problems  with a non 
> compliant SIP device, so this could well be a side effect of it.
>
> Any recommendation on tm memory usage ?
>
> opensips log
> ---
>
> Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: ERROR:tm:new_t: 
> out of mem
> Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]: 
> ERROR:tm:t_newtran: new_t failed
>
>  
>
> 
>
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Re: [OpenSIPS-Users] tm - module out of memory

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Brett,

in your case it was a temporary overload of the memory (due the looping) 
and not a leak, right ? - btw, it such cases, reducing the maxfwd value 
will help in reducing the number of loops. Also some script logic is 
good (to detect loops based on src ip, ruri, etc)

Regards,
Bogdan

Brett Nemeroff wrote:
> For what it's worth, I have also experienced out of memory error 
> occurring from large quantities of routing loops that I hadn't 
> detected. The memory loss was so bad that opensips stopped processing 
> calls (but never seemed to have crashed).
>
> Memory errors went away when I resolved the route loops. :)
> -Brett
>
>
> On Thu, Apr 29, 2010 at 7:22 PM, Julien Chavanton  > wrote:
>
>
> I found is a routing loop with another equipement, I was able to
> find some trace, this is not an Opensips related problem anymore.
>  
>
> 
> *From:* users-boun...@lists.opensips.org
>  on behalf of Julien
> Chavanton
> *Sent:* Thu 29/04/2010 11:22 PM
> *To:* OpenSIPS users mailling list; Users@lists.opensips.org
> 
> *Subject:* Re: [OpenSIPS-Users] tm - module out of memory
>
>  
> I have to diagnostic this system further, as before facing
> a memory shortage, there is a huge bunch of
> ERROR:tm:t_check: reply cannot be parsed
>  
>  
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH
> ROUTE**
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY
> ROUTE - td[10.1.16.50]**
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]:
> ERROR:tm:t_check: reply cannot be parsed
> Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST
> ROUTED ***
>
> 
> *From:* users-boun...@lists.opensips.org
>  on behalf of Julien
> Chavanton
> *Sent:* Thu 29/04/2010 8:27 PM
> *To:* Users@lists.opensips.org 
> *Subject:* [OpenSIPS-Users] tm - module out of memory
>
> tm module went out of memory, we did face other problems  with a
> non compliant SIP device, so this could well be a side effect of it.
>
> Any recommendation on tm memory usage ?
>
> opensips log
> 
> ---
>
> Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]:
> ERROR:tm:new_t: out of mem
> Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]:
> ERROR:tm:t_newtran: new_t failed
>
>  
>
>
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>
>
> 
>
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Re: [OpenSIPS-Users] RADIUS AVP related Error

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Vishu,

have you defined the Bcontact radius AVP in the dictionary ? what value 
did you configure for it ?

Regards,
Bogdan

vishu gaddi wrote:
> I have configured Opensips, Radius and CDRTool. But while making a
> call i am getting following error and no Data is inserted into
> 'radacct'  table.
>
> ERROR =>
> rc_avpair_new: unknown attribute 0
> Apr 29 06:19:09 opensips /sbin/opensips[32327]:
> ERROR:aaa_radius:rad_avp_add: failure
> Apr 29 06:19:09 opensips /sbin/opensips[32327]:
> ERROR:acc:acc_aaa_request: failed to add Bcontact, 10
>
>
> Please help me, where should i look for ?
>
> Thanks,
> Vishu
>
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Re: [OpenSIPS-Users] opensipsctl erros

2010-05-03 Thread Bogdan-Andrei Iancu
Hi guys,

the Airton's report may be right - the correct operator for string 
comps  is "==" and not "-eq" (which is used for arithmetic operands):
   http://tldp.org/LDP/abs/html/comparison-ops.html

So, we might need to change the script to be on the safe side.

Regards,
Bogdan

Vadim Grinco wrote:
> Hi,
>
> On Fri, Apr 30, 2010 at 4:32 PM, brianpocock 
>  > wrote:
>
>
> Just checked my configuration and it works fine for me with:
> if [ $DBENGINE -eq "MYSQL" ]
> when doing opensipsctl alias_db list/show
>
> Not sure whether you made a typo with:
> if [ $DBENGINE -eq "MYSQL" ]
> if [ $DBENGINE -eq "PGSQL" ]
>
> but it should read:
> if [ $DBENGINE -eq "MYSQL" ]
> elif [ $DBENGINE -eq "PGSQL" ]
>
> I don't see any substantial difference between if and elif in this 
> situation. Yes, it will save you a couple of milliseconds to execute 
> because it won't do the second comparison, but the logyc is the same.
> In this case the operator is incorrect. If you man sh, you will see 
> that -eq is an arithmetic binary operator, but = or == is used to 
> compare strings.
>
> You might have an interpreter which will understand both versions, but 
> don't forget there're lots of them out there.
>
>  
>
> Hope this helps
> --
> View this message in context:
> 
> http://opensips-open-sip-server.1449251.n2.nabble.com/opensipsctl-erros-tp4985896p4985961.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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>
>
>
> -- 
> Best regards,
> Vadim Grinco
>
> 
>
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Re: [OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.conf doesn't exist

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Vishu,

The client.conf dule is provided by freeradius installation .

Change in opensips cfg the aaa_url to point to the actual location where 
the client.conf was installed on your system.

Regards,
Bogdan

vishu gaddi wrote:
> I have installed opensips 1.6, CDRTool 7.1.1 and RADIUS 1.1.7
>
>
> To enable Opensips to send accounting logs to Freeradius, there should 
> be path defined in opensips.cfg as per documentation  as below:
>
> *modparam("acc", "aaa_url","radius:/etc/opensips/radius/client.conf")*
>
>
>
> But there doesn't exist */etc/opensips/radius/client.conf* file, SO 
> which path i should define here.. i am suspected that my opensips is 
> not compiled with radius properly.. is it ?
>
> Pls suggest..
> 
>
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Re: [OpenSIPS-Users] Trunking Calls Onward

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Mike,

I see several solutions already posted, but I may suggest one ever 
simpler and much faster :
- the ENUM have the provisioning overhead  (to set each DID ) and 
the penalty on lookup (dns lookup for the calls)
- the alias_db_lookup suffers from the same issues (provisioning 
each DID and DB query for each call)

So, I suggest using the dialplan module for this purpose 
(http://www.opensips.org/html/docs/modules/1.6.x/dialplan.html). The 
module allows you do the translation based on regexp, so you can define 
blocks of DID (regexp based matching) to lead to a specific domain / ip 
(where the forward the call).

See 
http://www.opensips.org/html/docs/modules/1.6.x/dialplan.html#id227206 , 
case 1.4.1.2 where you can use a regxep to match the input (the DID) and 
return a fix string, a replacement (the target domain).

So, you can provision blocks of DIDs (via regexps), use opensips Control 
Panel to provision the rule (see .http://opensips-cp.sourceforge.net/)  
Also the module is doing caching, so the lookup is really fast.

Regards,
Bogdan

Mike O'Connor wrote:
> Hi All
>
> I have a need to forward calls onward for a range of DID's, but the
> other end is not going to Register. I think this is called trunking.
>
> I need to be able to configure the DID's and the ip/port there being on
> forwarded too.
> What methods should I use to do this ?
>
> I've look at a number of options but the issue is that for the functions
> like 'seturi' do not allow variables only static strings.
>
> Thanks
> Mike
>
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Re: [OpenSIPS-Users] Advice on OpenSIPS capabilities

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Dennis,

You should consider the case where opensips is advertising a public IP 
(instead of its own private one ) when sending calls to public internet 
- see advertise_address param 
(http://www.opensips.org/Resources/DocsCoreFcn16#toc24).

Regards,
Bogdan

Dennis Cartier wrote:
> I need some advice on how suitable OpenSIPS would be for solving a 
> situation that I am struggling with.
>
> I have a VOIP system (Cisco UC520) behind NAT on my LAN. I need to 
> setup a couple of extensions that are on the public Internet (Asterisk 
> based). The issue that I am trying to deal with is that the local VOIP 
> system sends INVITES with a private IP as it is unaware of what the 
> true public IP is.
>
> In my search for a way to dela with this I ran across OpenSIPS. I am 
> wondering if I can use OpenSIPS as a proxy and/or could I use it to 
> re-write the IP in the SIP packets to the proper public IP?
>
> Any advice you can give as to how applicable OpenSIPS is to this task 
> and if so, which function of OpenSIPS would be best suited to this 
> problem.
>
> Thanks,
>
> Dennis
>
>
> -- 
> “The music business is a cruel and shallow money trench, a long 
> plastic hallway where thieves and pimps run free, and good men die 
> like dogs. There’s also a negative side. ”
>
> Hunter S. Thompson, US journalist (1939 – 2005)
>
> 
>
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Re: [OpenSIPS-Users] Opensips 1.6 : /etc/opensips/radius/clients.confdoesn't exist

2010-05-03 Thread Bogdan-Andrei Iancu
Better define the the aaa_url param in opensips to point to 
*/etc/freeradius/clients.conf
*
Regards,
Bogdan

vishu wrote:
>
> So should I copy clients.conf from */etc/freeradius/clients.conf* to 
> */etc/opensips/radisus/client.conf*  or can define the path 
> /etc/freeradius/clients.conf in opensips.conf , does it matter ?
>
> 
>
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Re: [OpenSIPS-Users] Need help on realtime integration OpenSIPS with Freeswitch.

2010-05-03 Thread Bogdan-Andrei Iancu
Hi,

what is the purpose of the integration? FS should be a media server (VM, 
conf, etc) or a PBX, etc ?

Regards,
Bogdan

hung nguyen wrote:
> Hi list.
> I read some document about interation OpenSIPS with Asterisk. But now, 
> i need to deploy with Freeswitch. Anybody done this?
> I deploy Freeswitch with using ODBC in the core, but i don't know how 
> to create view share datebase between FS DB and Opensips DB.
> Tks.
> 
>
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Re: [OpenSIPS-Users] Opensips logging issue?

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Adelson,

it is classic problem - if you have to much debugging (like debug=4) , 
opensips will become slow due to syslog interaction (more or less syslog 
will drag opensips down).

For this reason I do not recommend to run in a production env with 
debug>=4 level.

Also check your syslog conf - you may gain some speed up if you set 
async writing into log file (see the ~ or - prefix to the log file in 
syslog.conf). Also man page for syslog will explain you better this ;)

Regards,
Bogdan

Adelson O. Junior wrote:
> Hello list,
>
> We were used to disable logs (debug=0) due to problems regarding to 
> old Openser versions, and this became standard when we were going to 
> migrate from Test to Production environments.
> But in the last one we forget this setting, and the log was enabled in 
> Production Env, but no one was concerned about this, we though that 
> this problem didn' t happen in Opensips, but it did.
>
> So, the problem itself is:
> The Opensips receive a INVITE packet, and took too long to forward.
>
> Looking in the log, we could see the lines about the processing of 
> INVITE (instantly forwarded). But the time between the packet arrival 
> (capturing by ngrep) and your lines in the log its about  2, 3, 4 
> seconds of difference.
>
> This problem, as I said we already faced it on openser versions, and 
> we only have disable logs (debug=0) in opensips.cfg, restart the 
> service and it got back to work normally.
>
> Did someone faced this problem?
>
> -- 
> Adelson
> 
>
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Re: [OpenSIPS-Users] Modify Invite

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Brad,

so printing the $ru after the changes (with xlog) shows the right value, 
but in the outgoing INVITE you do not see that value ??  If so, maybe 
you have a second change on RURI that is changing your change :).

try placing xlogs for $ru in different location in the code between your 
change and the t_relay() to see if the $ru gets changes on the way.

Regards,
Bogdan

Brad Bendy wrote:
> Hi,
>
> Im bringing this back up again.
>
> Ive had to switch our INVITE format as some upstreams parse the INVITE 
> differently, I just need to change the order of variables around.
>
> Now the weird thing is, I set the $ru and then when I check to see 
> what $ru is set to, it's been set correct, the order and all variables 
> are correct, but then openSIPs sends the INVITE out it fails to modify 
> it at all.
>
> I was using:
> $ru='sip:' + $rU + '@' + $rd + ';npdi=yes' + ';rn=' + $avp(s:lrn);
>
> now
> $ru='sip:' + $rU + ';npdi=yes' + ';rn=+1' + $avp(s:lrn) + '@' + $rd;
>
> Before the INVITE would look like:
>
> INVITE sip:+112355599...@1.1.1.1*;npdi=yes;rn=+1123555 SIP/2.0
>
> but now it should look like INVITE 
> sip:+1123555;npdi=yes;rn=+112355599...@1.1.1.1* SIP/2.0
>
> it currently looks like INVITE sip:+11235559...@*1.1.1.1* SIP/2.0.
>
> Any ideals why just the positions of the AVPs and the other text would 
> matter? I think it's weird when I use xlog() to print what $ru is set 
> to it shows the correct value just OpenSIPs does not modify the 
> INVITE. Not sure if anyone else has ran into this on the list or not.
>
> Thanks!
>
>
> On Mon, 2010-02-22 at 13:21 -0700, Brad Bendy wrote:
>> I just did:
>> $ru='sip:' + $rU + '@' + $rd + ';npdi=yes' + ';rn=' + $avp(s:lrn);
>>
>> That works like a champ, thanks for help on this, rewriting the ru 
>> direct took care of it!
>>
>> Thanks again
>>
>> On Mon, 2010-02-22 at 18:24 +0200, Bogdan-Andrei Iancu wrote:
>>> Brad,
>>>
>>> the function does not support variables as parameters :(.
>>>
>>> But a simple workaround, if you want to add a parameter to the RURI is:
>>> $ru = $ru + ";rn=" + $avp(s:foo) ;
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Brad Bendy wrote:
>>> > Hi Bogdan,
>>> >
>>> > I just tried the add_uri_param() but when a avp since I need to pass a 
>>> > dynamic value to it, and it errors out with a syntax error. If I try 
>>> > with double quotes around the avp it then literally displays 
>>> > "$avp(s:foo)" when it adds to the URI.
>>> >
>>> > Something im missing?
>>> >
>>> > Thanks
>>> >
>>> > On Mon, 2010-02-22 at 10:45 +0200, Bogdan-Andrei Iancu wrote:
>>> >> Hi Brad,
>>> >>
>>> >> Maybe http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id270649 ?
>>> >>
>>> >> Regards,
>>> >> Bogdan
>>> >>
>>> >> Brad Bendy wrote:
>>> >> > Hi list,
>>> >> >
>>> >> > I need to add npdi=yes;rn=xx when I send a INVITE out. Ive 
>>> >> > looked and don't really see any way to do this exactly. Is their a 
>>> >> > module im missing to do this, or some sort of method? I thought this 
>>> >> > would be in textops maybe but I do not see it in that module either.
>>> >> >
>>> >> > Any help or pointers would be great.
>>> >> >
>>> >> > Thanks!
>>> >> > 
>>> >> >


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Re: [OpenSIPS-Users] ERROR:perl:parser_init: failed to load perl file

2010-05-03 Thread samoh

Hi Vadim,

there is a result of strace command :

r...@samy-desktop:~# strace file "/home/opensips/etc/opensips/script.pl"
execve("/usr/bin/file", ["file", "/home/opensips/etc/opensips/scri"...], [/*
21 vars */]) = 0
brk(0)  = 0x87ca000
access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
directory)
mmap2(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb7888000
access("/etc/ld.so.preload", R_OK)  = -1 ENOENT (No such file or
directory)
open("/etc/ld.so.cache", O_RDONLY)  = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=55457, ...}) = 0
mmap2(NULL, 55457, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb787a000
close(3)= 0
access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
directory)
open("/usr/lib/libmagic.so.1", O_RDONLY) = 3
read(3,
"\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0P!\0\0004\0\0\0H"..., 512) =
512
fstat64(3, {st_mode=S_IFREG|0644, st_size=87680, ...}) = 0
mmap2(NULL, 90812, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) =
0xb7863000
mmap2(0xb7878000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x14) = 0xb7878000
close(3)= 0
access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
directory)
open("/lib/libz.so.1", O_RDONLY)= 3
read(3,
"\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0P\31\0\0004\0\0\0\0"...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=83552, ...}) = 0
mmap2(NULL, 86284, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) =
0xb784d000
mmap2(0xb7861000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x13) = 0xb7861000
close(3)= 0
access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
directory)
open("/lib/tls/i686/cmov/libc.so.6", O_RDONLY) = 3
read(3,
"\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320h\1\0004\0\0\0\344"...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0755, st_size=1442180, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb784c000
mmap2(NULL, 1451632, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) =
0xb76e9000
mprotect(0xb7845000, 4096, PROT_NONE)   = 0
mmap2(0xb7846000, 12288, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x15c) = 0xb7846000
mmap2(0xb7849000, 9840, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0xb7849000
close(3)= 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb76e8000
set_thread_area({entry_number:-1 -> 6, base_addr:0xb76e86c0, limit:1048575,
seg_32bit:1, contents:0, read_exec_only:0, limit_in_pages:1,
seg_not_present:0, useable:1}) = 0
open("/dev/urandom", O_RDONLY)  = 3
read(3, "\272\327\325"..., 3)   = 3
close(3)= 0
mprotect(0xb7846000, 8192, PROT_READ)   = 0
mprotect(0xb7861000, 4096, PROT_READ)   = 0
mprotect(0xb7878000, 4096, PROT_READ)   = 0
mprotect(0x804b000, 4096, PROT_READ)= 0
mprotect(0xb78a7000, 4096, PROT_READ)   = 0
munmap(0xb787a000, 55457)   = 0
brk(0)  = 0x87ca000
brk(0x87eb000)  = 0x87eb000
open("/usr/lib/locale/locale-archive", O_RDONLY|O_LARGEFILE) = -1 ENOENT (No
such file or directory)
open("/usr/share/locale/locale.alias", O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=2570, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb7887000
read(3, "# Locale name alias data base.\n# "..., 4096) = 2570
read(3, ""..., 4096)= 0
close(3)= 0
munmap(0xb7887000, 4096)= 0
open("/usr/lib/locale/fr_FR.UTF-8/LC_CTYPE", O_RDONLY) = -1 ENOENT (No such
file or directory)
open("/usr/lib/locale/fr_FR.utf8/LC_CTYPE", O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=256316, ...}) = 0
mmap2(NULL, 256316, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb76a9000
close(3)= 0
open("/usr/lib/gconv/gconv-modules.cache", O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=26048, ...}) = 0
mmap2(NULL, 26048, PROT_READ, MAP_SHARED, 3, 0) = 0xb7881000
close(3)= 0
stat64("/root/.magic", 0xbfae9c34)  = -1 ENOENT (No such file or
directory)
open("/etc/magic.mgc", O_RDONLY|O_LARGEFILE) = -1 ENOENT (No such file or
directory)
stat64("/etc/magic", {st_mode=S_IFREG|0644, st_size=111, ...}) = 0
open("/etc/magic", O_RDONLY|O_LARGEFILE) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=111, ...}) = 0
mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
0xb788
read(3, "# Magic local data for file(1) co"..., 4096) = 111
read(3, ""..., 4096)= 0
close(3)= 0
munmap(0xb788, 4096)= 0
open("/usr/share/file/magic.mgc", O_RDONLY|O_LARGEFILE) = 3
fstat64(3, {st_mode=S_IFREG|0

Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Bogdan-Andrei Iancu
Hi Ahmed,

as a hint, probably you do not handle correctly the case when only the 
callee is nated (caller is public) - for such cases, to see if rtpproxy 
is needed, after the lookup(location) the nat_bflag will will 
automatically set if the callee location is nated -> you can use that 
flag to detect the nated callee and to do the nat fixups -> force rtpp 
and fix the 200 ok from the callee (SDP and contact).

Regards,
Bogdan

Ahmed Munir wrote:
> Hi,
>
> Thanks for replying. Can you please check my configuration of OpenSIPs 
> what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.
>
> Please point out in which section do I required to add 
> force_rtp_proxy(), because I already configured Nat on it. kindly 
> advise me soon.
>
> On Fri, Apr 30, 2010 at 11:35 AM,  > wrote:
>
> Send Users mailing list submissions to
>users@lists.opensips.org 
>
> To subscribe or unsubscribe via the World Wide Web, visit
>http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> or, via email, send a message with subject or body 'help' to
>users-requ...@lists.opensips.org
> 
>
> You can reach the person managing the list at
>users-ow...@lists.opensips.org
> 
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Users digest..."
>
>
> Today's Topics:
>
>   1. Re: NAT Problem using Nat helper (Laszlo)
>
>
> --
>
> Message: 1
> Date: Fri, 30 Apr 2010 08:35:00 +0200
> From: Laszlo mailto:las...@voipfreak.net>>
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list  >
> Message-ID:
>  
>   >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Ahmed,
>
> As you can see, the other party gets local ip in SDP
>
> c=IN IP4 192.168.0.168.
>
> You can try to play with flags:
> http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028
>
> -Laszlo
>
>
>
> 2010/4/30 Ahmed Munir  >
>
> >
> >
> > Hi.
> >
> > Thanks for your reply, the traces are metioned below;
> >
> > U 203.215.176.22:55134  ->
> 11.22.33.44:5060 
> > .
> > .
> > ..
> >
> > U 81.201.82.45:5060  ->
> 11.22.33.44:5060 
> > INVITE sip:1234...@11.22.33.44
>   > SIP/2.0.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
> .
> > CSeq: 102 INVITE.
> > From: "4572727220"   >
> > >;tag=43772.
> > To: mailto:sip%3a1234...@11.22.33.44>
> mailto:sip%253a1234...@11.22.33.44>>>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060 
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0.
> > Max-Forwards: 69.
> > Content-Type: application/sdp.
> > Contact: .
> > User-Agent: Vox Callcontrol.
> > Content-Length: 210.
> > .
> > v=0.
> > o=root 13293 13293 IN IP4 81.201.82.146.
> > s=session.
> > c=IN IP4 81.201.82.146.
> > t=0 0.
> > m=audio 11458 RTP/AVP 8 0.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=silenceSupp:off - - - -.
> > a=ptime:20.
> > a=sendrecv.
> >
> >
> > U 11.22.33.44:5060  ->
> 81.201.82.45:5060 
> > SIP/2.0 100 Giving a try.
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
> .
> > CSeq: 102 INVITE.
> > From: "4572727220"   >
> > >;tag=43772.
> > To: mailto:sip%3a1234...@11.22.33.44>
> mailto:sip%253a1234...@11.22.33.44>>>.
> > Via: SIP/2.0/UDP 81.201.82.45:5060 
> > ;branch=z9hG4bKdb42364564e21b159baaa8a741307ca0;rport=5060.
> > Server: OpenSIPS (1.6.1-notls (i386/linux)).
> > Content-Length: 0.
> > .
> >
> >
> > U 11.22.33.44:5060  ->
> 203.215.176.22:55134 
> > INVITE sip:4...@203.215.176.22:55134;rinstance=25bfe05618433c26
> SIP/2.0.
> > Record-Route: .
> > Call-ID: nmxrpjipejfwxkqq3a75zd3...@81.201.82.45
> .
> 

[OpenSIPS-Users] OpenSIPS failover

2010-05-03 Thread rajib deka
Hi List,

Cn some some please explain me how OpenSIPS supports HA and redundancy? I
have not found anything on this on the site. I have an implementation with
virtual ip for HA and its working fine. I just want to know is there any
internal module or configuration with OpenSIPS using which we can achieve HA
and redundancy.

-- 
Rajib Deka
Software Engineer
Servion Global Solution
Chennai, India

Mobile No: + 91 80157 09130
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Re: [OpenSIPS-Users] Need help on realtime integration OpenSIPS with Freeswitch.

2010-05-03 Thread hung nguyen
Hi.

I deploy cluster Freeswitch boxes with share Database( use ODBC),they act as
PBX (register server, proxy server, media server ...).
Now FS is load balanced via DNS-SRV record.

I read in wiki about: Load balance using OpenSIPS,Realtime integration
OpenSIPS with Asterisk; and think it can be apply with FS.
I want to deploy Opensips between FS boxes, all users and gateways.
Opensips will act as register server for all users, and ( if can ) load
balance server for all FS boxes.

Thanks to suggestions.
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[OpenSIPS-Users] RADIUS AVP related Error

2010-05-03 Thread vishu gaddi
I have not defined Bcontact anywhere..

*Here is my opensips.cfg*


modparam("acc", "aaa_extra",   "Digest-User-Name=$Au; \
   Calling-Station-Id=$from; \
   Called-Station-Id=$to; \
   Sip-Translated-Request-URI=$ru; \
   Sip-RPid=$avp(s:rpid); \
   Source-IP=$avp(s:source_ip); \
   Source-Port=$avp(s:source_port); \
   SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
   Canonical-URI=$avp(s:can_uri); \
   Billing-Party=$avp(billing_party); \
   Divert-Reason=$avp(s:divert_reason);
\
   User-Agent=$hdr(user-agent); \
   Contact=$hdr(contact); \
   Event=$hdr(event); \
   ENUM-TLD=$avp(s:enum_tld)")
modparam("acc", "aaa_extra","via=$hdr(Via[*]); email=$avp(s:email);
Bcontact=$ct / reply")




*and here is dictionary.opensips*

### SIP Attributes ###
ATTRIBUTE Sip-Method   101  integer# Schulzrinne, acc
ATTRIBUTE Sip-Response-Code102  integer# Schulzrinne, acc
ATTRIBUTE Sip-Cseq 103  string # Schulzrinne, acc
ATTRIBUTE Sip-To-Tag   104  string # Schulzrinne, acc
ATTRIBUTE Sip-From-Tag 105  string # Schulzrinne, acc
ATTRIBUTE Sip-Branch-ID106  string
ATTRIBUTE Sip-Translated-Request-URI   107  string # Proprietary, acc
ATTRIBUTE Sip-Uri-User 208  string # Proprietary,
auth_radius
ATTRIBUTE Sip-Group211  string # Proprietary,
group_radius
ATTRIBUTE Sip-Rpid 213  string # Proprietary,
auth_radius
ATTRIBUTE Billing-Party218  string
ATTRIBUTE SIP-AVP  225  string # Proprietary,
avp_radius

### Sip-Method Values ###
VALUE Sip-Method Undefined  0
VALUE Sip-Method Invite 1
VALUE Sip-Method Cancel 2
VALUE Sip-Method Ack4
VALUE Sip-Method Bye8
VALUE Sip-Method Info   16
VALUE Sip-Method Options32
VALUE Sip-Method Update 64
VALUE Sip-Method Register   128
VALUE Sip-Method Message256
VALUE Sip-Method Subscribe  512
VALUE Sip-Method Notify 1024
VALUE Sip-Method Prack  2048
VALUE Sip-Method Refer  4096
VALUE Sip-Method Publish8192
VALUE Sip-Method Other  16384

### Sip-Response-Code Values ###
VALUE Sip-Response-Code  Undefined  0
VALUE Sip-Response-Code  Invite 1
VALUE Sip-Response-Code  Cancel 2
VALUE Sip-Response-Code  Ack4
VALUE Sip-Response-Code  Bye8
VALUE Sip-Response-Code  Info   16
VALUE Sip-Response-Code  Options32
VALUE Sip-Response-Code  Update 64
VALUE Sip-Response-Code  Register   128
VALUE Sip-Response-Code  Message256
VALUE Sip-Response-Code  Subscribe  512
VALUE Sip-Response-Code  Notify 1024
VALUE Sip-Response-Code  Prack  2048
VALUE Sip-Response-Code  Refer  4096
VALUE Sip-Response-Code  Publish8192
VALUE Sip-Response-Code  Other  16384

### Acct-Status-Type Values ###
VALUE Acct-Status-Type Start 1 # RFC2866, acc
VALUE Acct-Status-Type Stop  2 # RFC2866, acc
VALUE Acct-Status-Type Failed   15 # RFC2866, acc

### Service-Type Values ###
VALUE Service-Type Call-Check   10 # RFC2865, uri_radius
VALUE Service-Type Group-Check  12 # Proprietary,
group_radius
VALUE Service-Type Sip-Session  15 # Schulzrinne, acc,
auth_radius
VALUE Service-Type SIP-Caller-AVPs  30 # Proprietary,
avp_radius
VALUE Service-Type SIP-Callee-AVPs  31 # Proprietary,
avp_radius

### Attributes added by AG Projects ###
ATTRIBUTE   Source-IP  214 string
ATTRIBUTE   Source-Port215 string
ATTRIBUTE   Canonical-URI  216 string
ATTRIBUTE   Delay-Time 217 string
ATTRIBUTE   Divert-Reason  219 string
ATTRIBUTE   X-RTP-Stat 220 string
ATTRIBUTE   From-Header221 string
ATTRIBUTE   User-Agent 222 string
ATTRIBUTE   Contact223 string
ATTRIBUTE   Event  224 string
#ATTRIBUTE   Event-Timestamp   230 string
ATTRIBUTE   SIP-Proxy-IP   231 string
ATTRIBUTE   ENUM-TLD   2

Re: [OpenSIPS-Users] ERROR:perl:parser_init: failed to load perl file

2010-05-03 Thread Vadim Grinco
Yes, opensips can read the file, and the problem is not in file permissions
or selinux.


On Mon, May 3, 2010 at 11:35 AM, samoh  wrote:

>
> Hi Vadim,
>
> there is a result of strace command :
>
> r...@samy-desktop:~# strace file "/home/opensips/etc/opensips/script.pl"
> execve("/usr/bin/file", ["file", "/home/opensips/etc/opensips/scri"...],
> [/*
> 21 vars */]) = 0
> brk(0)  = 0x87ca000
> access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
> directory)
> mmap2(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
> 0xb7888000
> access("/etc/ld.so.preload", R_OK)  = -1 ENOENT (No such file or
> directory)
> open("/etc/ld.so.cache", O_RDONLY)  = 3
> fstat64(3, {st_mode=S_IFREG|0644, st_size=55457, ...}) = 0
> mmap2(NULL, 55457, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb787a000
> close(3)= 0
> access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
> directory)
> open("/usr/lib/libmagic.so.1", O_RDONLY) = 3
> read(3,
> "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0P!\0\0004\0\0\0H"..., 512)
> =
> 512
> fstat64(3, {st_mode=S_IFREG|0644, st_size=87680, ...}) = 0
> mmap2(NULL, 90812, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) =
> 0xb7863000
> mmap2(0xb7878000, 8192, PROT_READ|PROT_WRITE,
> MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x14) = 0xb7878000
> close(3)= 0
> access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
> directory)
> open("/lib/libz.so.1", O_RDONLY)= 3
> read(3,
> "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0P\31\0\0004\0\0\0\0"...,
> 512) = 512
> fstat64(3, {st_mode=S_IFREG|0644, st_size=83552, ...}) = 0
> mmap2(NULL, 86284, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) =
> 0xb784d000
> mmap2(0xb7861000, 8192, PROT_READ|PROT_WRITE,
> MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x13) = 0xb7861000
> close(3)= 0
> access("/etc/ld.so.nohwcap", F_OK)  = -1 ENOENT (No such file or
> directory)
> open("/lib/tls/i686/cmov/libc.so.6", O_RDONLY) = 3
> read(3,
> "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\320h\1\0004\0\0\0\344"...,
> 512) = 512
> fstat64(3, {st_mode=S_IFREG|0755, st_size=1442180, ...}) = 0
> mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
> 0xb784c000
> mmap2(NULL, 1451632, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0)
> =
> 0xb76e9000
> mprotect(0xb7845000, 4096, PROT_NONE)   = 0
> mmap2(0xb7846000, 12288, PROT_READ|PROT_WRITE,
> MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x15c) = 0xb7846000
> mmap2(0xb7849000, 9840, PROT_READ|PROT_WRITE,
> MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0xb7849000
> close(3)= 0
> mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
> 0xb76e8000
> set_thread_area({entry_number:-1 -> 6, base_addr:0xb76e86c0, limit:1048575,
> seg_32bit:1, contents:0, read_exec_only:0, limit_in_pages:1,
> seg_not_present:0, useable:1}) = 0
> open("/dev/urandom", O_RDONLY)  = 3
> read(3, "\272\327\325"..., 3)   = 3
> close(3)= 0
> mprotect(0xb7846000, 8192, PROT_READ)   = 0
> mprotect(0xb7861000, 4096, PROT_READ)   = 0
> mprotect(0xb7878000, 4096, PROT_READ)   = 0
> mprotect(0x804b000, 4096, PROT_READ)= 0
> mprotect(0xb78a7000, 4096, PROT_READ)   = 0
> munmap(0xb787a000, 55457)   = 0
> brk(0)  = 0x87ca000
> brk(0x87eb000)  = 0x87eb000
> open("/usr/lib/locale/locale-archive", O_RDONLY|O_LARGEFILE) = -1 ENOENT
> (No
> such file or directory)
> open("/usr/share/locale/locale.alias", O_RDONLY) = 3
> fstat64(3, {st_mode=S_IFREG|0644, st_size=2570, ...}) = 0
> mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) =
> 0xb7887000
> read(3, "# Locale name alias data base.\n# "..., 4096) = 2570
> read(3, ""..., 4096)= 0
> close(3)= 0
> munmap(0xb7887000, 4096)= 0
> open("/usr/lib/locale/fr_FR.UTF-8/LC_CTYPE", O_RDONLY) = -1 ENOENT (No such
> file or directory)
> open("/usr/lib/locale/fr_FR.utf8/LC_CTYPE", O_RDONLY) = 3
> fstat64(3, {st_mode=S_IFREG|0644, st_size=256316, ...}) = 0
> mmap2(NULL, 256316, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb76a9000
> close(3)= 0
> open("/usr/lib/gconv/gconv-modules.cache", O_RDONLY) = 3
> fstat64(3, {st_mode=S_IFREG|0644, st_size=26048, ...}) = 0
> mmap2(NULL, 26048, PROT_READ, MAP_SHARED, 3, 0) = 0xb7881000
> close(3)= 0
> stat64("/root/.magic", 0xbfae9c34)  = -1 ENOENT (No such file or
> directory)
> open("/etc/magic.mgc", O_RDONLY|O_LARGEFILE) = -1 ENOENT (No such file or
> directory)
> stat64("/etc/magic", {st_mode=S_IFREG|0644, st_size=111, ...}) = 0
> open("/etc/magic", O_RDONLY|O_LARGEFILE) = 3
> fstat64(3, {st_mode=S_IFREG|0644, st_size=111, ...}) = 0
> mmap2(NULL, 4096,

Re: [OpenSIPS-Users] tm - module out of memory

2010-05-03 Thread Brett Nemeroff
Bogdan,
Those are good ideas, I'll give it a shot.. I think the memory loss was
temp, but the biggest affect of it was that the fifo process died entirely;
requiring a restart to have fifo capabilities.
-Brett


On Mon, May 3, 2010 at 3:46 AM, Bogdan-Andrei Iancu
wrote:

> Hi Brett,
>
> in your case it was a temporary overload of the memory (due the looping)
> and not a leak, right ? - btw, it such cases, reducing the maxfwd value
> will help in reducing the number of loops. Also some script logic is
> good (to detect loops based on src ip, ruri, etc)
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
> > For what it's worth, I have also experienced out of memory error
> > occurring from large quantities of routing loops that I hadn't
> > detected. The memory loss was so bad that opensips stopped processing
> > calls (but never seemed to have crashed).
> >
> > Memory errors went away when I resolved the route loops. :)
> > -Brett
> >
> >
> > On Thu, Apr 29, 2010 at 7:22 PM, Julien Chavanton  > > wrote:
> >
> >
> > I found is a routing loop with another equipement, I was able to
> > find some trace, this is not an Opensips related problem anymore.
> >
> >
> >
> 
> > *From:* users-boun...@lists.opensips.org
> >  on behalf of Julien
> > Chavanton
> > *Sent:* Thu 29/04/2010 11:22 PM
> > *To:* OpenSIPS users mailling list; Users@lists.opensips.org
> > 
> > *Subject:* Re: [OpenSIPS-Users] tm - module out of memory
> >
> >
> > I have to diagnostic this system further, as before facing
> > a memory shortage, there is a huge bunch of
> > ERROR:tm:t_check: reply cannot be parsed
> >
> >
> > Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***BRANCH
> > ROUTE**
> > Apr 29 19:16:58 osip /usr/local/sbin/opensips[20104]: ***REPLY
> > ROUTE - td[10.1.16.50]**
> > Apr 29 19:16:58 osip /usr/local/sbin/opensips[20103]:
> > ERROR:tm:t_check: reply cannot be parsed
> > Apr 29 19:16:58 osip /usr/local/sbin/opensips[20101]: ***HOST
> > ROUTED ***
> >
> >
> 
> > *From:* users-boun...@lists.opensips.org
> >  on behalf of Julien
> > Chavanton
> > *Sent:* Thu 29/04/2010 8:27 PM
> > *To:* Users@lists.opensips.org 
> > *Subject:* [OpenSIPS-Users] tm - module out of memory
> >
> > tm module went out of memory, we did face other problems  with a
> > non compliant SIP device, so this could well be a side effect of it.
> >
> > Any recommendation on tm memory usage ?
> >
> > opensips log
> >
> ---
> >
> > Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]:
> > ERROR:tm:new_t: out of mem
> > Apr 29 19:17:33 osip /usr/local/sbin/opensips[20101]:
> > ERROR:tm:t_newtran: new_t failed
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
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>
>
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Re: [OpenSIPS-Users] NAT Problem using Nat helper

2010-05-03 Thread Ahmed Munir
Hi,

Thanks for supporting me, really appreciated your help.


> Date: Mon, 03 May 2010 12:39:55 +0300
> From: Bogdan-Andrei Iancu 
> Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> To: OpenSIPS users mailling list 
> Message-ID: <4bde99eb.9090...@voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed,
>
> as a hint, probably you do not handle correctly the case when only the
> callee is nated (caller is public) - for such cases, to see if rtpproxy
> is needed, after the lookup(location) the nat_bflag will will
> automatically set if the callee location is nated -> you can use that
> flag to detect the nated callee and to do the nat fixups -> force rtpp
> and fix the 200 ok from the callee (SDP and contact).
>
> Regards,
> Bogdan
>
> Ahmed Munir wrote:
> > Hi,
> >
> > Thanks for replying. Can you please check my configuration of OpenSIPs
> > what I posted on my previous digest i.e.Users Digest, Vol 21, Issue 146.
> >
> > Please point out in which section do I required to add
> > force_rtp_proxy(), because I already configured Nat on it. kindly
> > advise me soon.
> >
> > On Fri, Apr 30, 2010 at 11:35 AM,  > > wrote:
> >
> > Send Users mailing list submissions to
> >users@lists.opensips.org 
> >
> > To subscribe or unsubscribe via the World Wide Web, visit
> >http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > or, via email, send a message with subject or body 'help' to
> >users-requ...@lists.opensips.org
> > 
> >
> > You can reach the person managing the list at
> >users-ow...@lists.opensips.org
> > 
> >
> > When replying, please edit your Subject line so it is more specific
> > than "Re: Contents of Users digest..."
> >
> >
> > Today's Topics:
> >
> >   1. Re: NAT Problem using Nat helper (Laszlo)
> >
> >
> >
> --
> >
> > Message: 1
> > Date: Fri, 30 Apr 2010 08:35:00 +0200
> > From: Laszlo mailto:las...@voipfreak.net>>
> > Subject: Re: [OpenSIPS-Users] NAT Problem using Nat helper
> > To: OpenSIPS users mailling list  > >
> > Message-ID:
> >
> >   >  r2za950b2031004292335re52872d5tf60c9830c8c21...@mail.gmail.com>>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Hi Ahmed,
> >
> > As you can see, the other party gets local ip in SDP
> >
> > c=IN IP4 192.168.0.168.
> >
> > You can try to play with flags:
> >
> http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028
> >
> > -Laszlo
> >
> >
> >
> >
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
>
>
> --
>
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>
> End of Users Digest, Vol 22, Issue 13
> *
>



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Ahmed Munir
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Re: [OpenSIPS-Users] ERROR:perl:parser_init: failed to load perl file

2010-05-03 Thread samoh

I finally found where was the error.
the header of my perl file (#/usr/bin/perl) was incorrect, so, to resolve
the probleme I added the line 
modparam("perl", "modpath", "/home/opensips/lib/opensips/perl") 

Thanks for all.

Sam.
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Re: [OpenSIPS-Users] dialog module

2010-05-03 Thread wüber

Hello Richard,

I've configured my system as you suggested and now going on with my setup
and application, I have a similar problem with the CANCEL message.
If a make a call and, before answering the call, I close it from the caller
side, sending a CANCEL message, I get lots of CANCEL packets and ACK packets
and again the message that the size is too big!
One of these CANCEL and ACK packets (the same I can see with wireshark) are
between the server and the end user, the other (that  do not see with
wireshark) are inside the Opensips server!

How I can solve this problem?
Thanks in advance.

Carmelo
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Re: [OpenSIPS-Users] Opensips logging issue?

2010-05-03 Thread Adelson O. Junior
Hey Bogdan,

thanks for the answer. I got it.
I will try syslog async on lab, decrease the debug level and see how it
behaves.

Adelson.



On Mon, May 3, 2010 at 6:27 AM, Bogdan-Andrei Iancu
wrote:

> Hi Adelson,
>
> it is classic problem - if you have to much debugging (like debug=4) ,
> opensips will become slow due to syslog interaction (more or less syslog
> will drag opensips down).
>
> For this reason I do not recommend to run in a production env with
> debug>=4 level.
>
> Also check your syslog conf - you may gain some speed up if you set
> async writing into log file (see the ~ or - prefix to the log file in
> syslog.conf). Also man page for syslog will explain you better this ;)
>
> Regards,
> Bogdan
>
> Adelson O. Junior wrote:
> > Hello list,
> >
> > We were used to disable logs (debug=0) due to problems regarding to
> > old Openser versions, and this became standard when we were going to
> > migrate from Test to Production environments.
> > But in the last one we forget this setting, and the log was enabled in
> > Production Env, but no one was concerned about this, we though that
> > this problem didn' t happen in Opensips, but it did.
> >
> > So, the problem itself is:
> > The Opensips receive a INVITE packet, and took too long to forward.
> >
> > Looking in the log, we could see the lines about the processing of
> > INVITE (instantly forwarded). But the time between the packet arrival
> > (capturing by ngrep) and your lines in the log its about  2, 3, 4
> > seconds of difference.
> >
> > This problem, as I said we already faced it on openser versions, and
> > we only have disable logs (debug=0) in opensips.cfg, restart the
> > service and it got back to work normally.
> >
> > Did someone faced this problem?
> >
> > --
> > Adelson
> > 
> >
> > ___
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> >
>
>
> --
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>
>
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Re: [OpenSIPS-Users] Are there in OpenSIPS modules like AGI on asterisk ?

2010-05-03 Thread Brett Nemeroff
Sam,
Not sure of your specific application, but the exec module has some pretty
serious performance problems. Actually, the problems arn't with the module,
but with what the module calls..

-Brett

On Mon, May 3, 2010 at 3:42 AM, samoh  wrote:

>
> Hi Bogdan,
>
> Thank you very much for your responses.
> I will tray to use exec module ;).
>
> Best regards.
> Sam.
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Re: [OpenSIPS-Users] Advice on OpenSIPS capabilities

2010-05-03 Thread Dennis Cartier
Thanks for the info Mike. Unfortunately the Asterisk box is not the end I
have control over. If we were using Asterisk on both ends (rather than
Cisco) we would have no issues.

Dennis

On Sun, May 2, 2010 at 8:11 PM, Mike O'Connor  wrote:

>  Hi Dennis
>
> It would seem to me that you should be able to handle this in Asterisk by
> configuring Asterisk to handle these calls via its internal NAT systems.
> Have you tried setting the sip.conf account to nat=yes for the calls from
> the Cisco UC520 ?
>
> Email me direct as it would be better to keep Asterisk related
> conversations of this list.
>
> Mike
>
>
> On 2/05/10 5:15 AM, Dennis Cartier wrote:
>
> I need some advice on how suitable OpenSIPS would be for solving a
> situation that I am struggling with.
>
> I have a VOIP system (Cisco UC520) behind NAT on my LAN. I need to setup a
> couple of extensions that are on the public Internet (Asterisk based). The
> issue that I am trying to deal with is that the local VOIP system sends
> INVITES with a private IP as it is unaware of what the true public IP is.
>
> In my search for a way to dela with this I ran across OpenSIPS. I am
> wondering if I can use OpenSIPS as a proxy and/or could I use it to re-write
> the IP in the SIP packets to the proper public IP?
>
> Any advice you can give as to how applicable OpenSIPS is to this task and
> if so, which function of OpenSIPS would be best suited to this problem.
>
> Thanks,
>
> Dennis
>
>
> --
> �The music business is a cruel and shallow money trench, a long plastic
> hallway where thieves and pimps run free, and good men die like dogs.
> There�s also a negative side. �
>
> Hunter S. Thompson, US journalist (1939 � 2005)
>
>
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>
>


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hallway where thieves and pimps run free, and good men die like dogs.
There’s also a negative side. ”

Hunter S. Thompson, US journalist (1939 – 2005)
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Re: [OpenSIPS-Users] dialog module

2010-05-03 Thread Richard Revels
There are several parts of the config where this type of problem can be checked 
and caught.  An easy one, provided you aren't running an asterisk on the same 
IP that talks to opensips or something, is this source ip and request domain 
check

if( $si == $rd )
{   
xlog("L_INFO", "Rejecting message that seems to be looping\n");
sl_send_reply( "483", "Something wrong in SIP Message - created 
loop" );
return(0);
}

Another thing to check in CANCEL and ACK handling is can message be routed 
somewhere from the message headers or is it part of an existing transaction

if( (!loose_route()) && (!t_check_trans()) )
{
if (isflagset(1))
xlog("L_INFO", "not forwarding $rm with 
debug turned on for $ci \n");
return(0);
}

If you have these and still have a problem it's going to require a lot more 
information about your config and some ngrep traces to nail down.  For me 
anyway.  Someone else might have some more thoughts just from past experience.

Richard

On May 3, 2010, at 9:40 AM, wüber wrote:

> 
> Hello Richard,
> 
> I've configured my system as you suggested and now going on with my setup
> and application, I have a similar problem with the CANCEL message.
> If a make a call and, before answering the call, I close it from the caller
> side, sending a CANCEL message, I get lots of CANCEL packets and ACK packets
> and again the message that the size is too big!
> One of these CANCEL and ACK packets (the same I can see with wireshark) are
> between the server and the end user, the other (that  do not see with
> wireshark) are inside the Opensips server!
> 
> How I can solve this problem?
> Thanks in advance.
> 
> Carmelo
> -- 
> View this message in context: 
> http://opensips-open-sip-server.1449251.n2.nabble.com/dialog-module-tp4967644p4997814.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
> 
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Re: [OpenSIPS-Users] Advice on OpenSIPS capabilities

2010-05-03 Thread Mike O'Connor
Hi Dennis

You would need to test this but it possible that opensips and mediaproxy
might help, as these two things setup correctly do rewrite SDP and SIP
packets to fix NAT issues.

Mike

On 4/05/10 3:17 AM, Dennis Cartier wrote:
> Thanks for the info Mike. Unfortunately the Asterisk box is not the
> end I have control over. If we were using Asterisk on both ends
> (rather than Cisco) we would have no issues.
>
> Dennis
>
> On Sun, May 2, 2010 at 8:11 PM, Mike O'Connor  > wrote:
>
> Hi Dennis
>
> It would seem to me that you should be able to handle this in
> Asterisk by configuring Asterisk to handle these calls via its
> internal NAT systems. Have you tried setting the sip.conf account
> to nat=yes for the calls from the Cisco UC520 ?
>
> Email me direct as it would be better to keep Asterisk related
> conversations of this list.
>
> Mike
>
>
> On 2/05/10 5:15 AM, Dennis Cartier wrote:
>> I need some advice on how suitable OpenSIPS would be for solving
>> a situation that I am struggling with.
>>
>> I have a VOIP system (Cisco UC520) behind NAT on my LAN. I need
>> to setup a couple of extensions that are on the public Internet
>> (Asterisk based). The issue that I am trying to deal with is that
>> the local VOIP system sends INVITES with a private IP as it is
>> unaware of what the true public IP is.
>>
>> In my search for a way to dela with this I ran across OpenSIPS. I
>> am wondering if I can use OpenSIPS as a proxy and/or could I use
>> it to re-write the IP in the SIP packets to the proper public IP?
>>
>> Any advice you can give as to how applicable OpenSIPS is to this
>> task and if so, which function of OpenSIPS would be best suited
>> to this problem.
>>
>> Thanks,
>>
>> Dennis
>>
>>
>> -- 
>> �The music business is a cruel and shallow money trench, a long
>> plastic hallway where thieves and pimps run free, and good men
>> die like dogs. There�s also a negative side. �
>>
>> Hunter S. Thompson, US journalist (1939 � 2005)
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> -- 
> “The music business is a cruel and shallow money trench, a long
> plastic hallway where thieves and pimps run free, and good men die
> like dogs. There’s also a negative side. ”
>
> Hunter S. Thompson, US journalist (1939 – 2005)
>

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