Re: [OpenSIPS-Users] (CDRTool) combining sip_trace/media records from multiple proxies

2010-11-15 Thread Adrian Georgescu 
Yes you can, just define the data sources for sip trace and media trace to 
point to the same databases.

Adrian

On Nov 11, 2010, at 12:38 PM, Jeff Pyle wrote:

> Hello,
> 
> In my network I have three Opensips instances routing traffic to, from and 
> across my network.  I use MySQL with radius for Opensips accounting, as well 
> as MySQL for sip_trace and MySQL + radius for Mediaproxy accouting.  I use 
> CDRTool to view the records together.
> 
> In CDRTool I have a separate data source for each proxy's records, as each is 
> written to a separate database.  That's fine.  This also leads to separate 
> sip_trace and mediaproxy records for the same call ID.
> 
> 1)  Is it possible to use the same sip_trace table for all three proxies, and 
> therefore view the traffic across all three proxies at the same time in 
> CDRTool no matter which proxy's traffic I choose to look at?  My hope here is 
> they will key on the same call ID.  All three proxies are VMs on the same 
> dom0 so their clocks are the same.
> 
> 2)  Mediaproxy is engaged only once across any of the three proxies for the 
> same call.  Can I use the same media_sessions table for all three dispatchers 
> to see the media records for a call no matter which proxy's traffic I'm 
> looking at?  The MySQL side seems easier here than the radius one.  I fear at 
> best I'd lose the view of the radius data unless I pick the "correct" proxy 
> in CDRTool.
> 
> My goal is to minimize the amount of places I have to look when looking for 
> the same data.
> 
> Any thoughts?  Am I trying to combine apples and pomegranates?
> 
> 
> - Jeff
> 
> ___
> 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

Re: [OpenSIPS-Users] need help on connection CDRTool to opensips

2010-11-15 Thread Tijmen de Mes 

Hi,



The following steps are performed to rate a CDR:

1. Determination of the billing party
a. SIP account u...@domain
b. SIP domain of the SIP account
c. Source IP of the session
   d. Default (when none of the above matches)
if there is no sip account u...@domain in cdrtool the cdrtool try next
option(sip domain) if sip domain does not exist it try next  
option(source ip)and

then the default option
please let me know if i am wrong

This is correct.


2. Determination of the destination id
a. CanonicalURI (the destination after all lookups inside the SIP  
Proxy)
b. SipTranslatedRequestURI (the Request URI as presented by the SIP  
UA)
c. CalledStationId (the content of the To header, used as a last  
resort)
if the CanonicalURI does not exist it try to next option(Request  
URI) ...

please let me know if i am wrong

This is also correct.


3. Determination of the costs
2 opensips2 is a gateway
   endpoint-opensips1---opensips2internet-VOIP  
service provider

So basically opensips2 is just a gateway to opensips1?



how can i config cdrtool rating all calls from opensips1 and opensips2
could you please guide me some steps to do

Did you load the sample data so you can see if a call gets rated?



i just try to see what happen in cdrtool, call from subscriber to  
subscriber


these are the logs
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime Duration=36000
callid=1289463574-3016-hiep...@192.168.1.36 From=sip:843...@192.168.1.39
Gateway=192.168.1.36 To=sip:00842...@192.168.1.39
Nov 11 12:39:46 opensips cdrtool[2302]: MaxSessionTime=unlimited  
Type=postpaid

callid=1289463574-3016-hiep...@192.168.1.36 billingparty=843...@192.168.1.39

log of call-control
Call id 1289463575-3016-hiep...@192.168.1.36 of 843...@192.168.1.39 to
sip:00842...@192.168.1.39 is postpaid not limited

it always return postpaid
You need to specify that a sip account is prepaid if you want it to  
act like a prepaid account.


--
Tijmen de Mes
AG Projects


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] loose_route()

2010-11-15 Thread Victor Gamov 

Thanks Bogdan!

On 11.11.2010 14:15, Bogdan-Andrei Iancu wrote:

Victor Gamov wrote:


I hope that loose_route() will process Route headers not RURI.

RFC-3261 16.4 says that
"The proxy MUST inspect the Request-URI of the request. If the
Request-URI of the request contains a value this proxy previously
placed into a Record-Route header field (see Section 16.6 item 4),
the proxy MUST replace the Request-URI in the request with the last
value from the Route header field, and remove that value from the
Route header field. The proxy MUST then proceed as if it received
this modified request."

So loose_route() inspect R-URI. But R-URI
sip:74951000...@x.x.x.x:5060
is not the same as inserted into Route
(sip:X.X.X.X;lr=on;ftag=2204003977).

A RR hdr cannot be completely copied to a RR, as syntactically speaking
they are different . RR is a name-addr spec, while RURI is a SIP
URI...what you see there as lr and ftag are RR hdr param, so cannot be
copied into RURI - just the URI part of RR is copied into RURI.


So Route header
Route: 
removed, R-URI untouched, next Route processed and $du is
sip:X.X.X.X:50080 now.


not really :
1) opensips finds its IP in RURI -> previous hop was a strict router
2) next hop is in the first Route hdr
3) next hop has "lr" param, so next hop is a loose router -> strict to
loose conversion
A) last Route is moved into RURI (end point address)
B) send (via $du) the call to next hop (from top most Route).


--
CU,
Victor Gamov
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] opensipsctl interaction

2010-11-15 Thread Anton Zagorskiy 
Hello.

I have some parameters that are stored in a DB. I'm loading it once time
when opensips starts.
After when they are changed in the DB I want to re-read it. How can I do
this without restarting opensips?






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru





___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Register attack!

2010-11-15 Thread Brett Nemeroff 
On Wed, Nov 3, 2010 at 12:23 PM, Flavio Goncalves wrote:

> Hi Saul,
>
> I did like your solution. My only concern about Pike was to block
> legitimate traffic. A SIP dialer can easily get to the pike threshold,
> but doing pike_check_req() just for register, options and bye requests
> seems to avoid this.
>
> The only "but" is,  the attack can also be done using INVITE and using
> Pike with INVITE can make you drop legitimate traffic, my initial
> concern. I think, that detecting authentication requests with wrong
> passwords or inexistent users is still the most generic solution. Just
> an opinion.
>
>
I personally just log each time there is an attempt from an unknown IP or
invalid user then just let fail2ban manage the threshold. Seems to work
pretty well.

The only thing that is really missing is a sort of system wide blacklist. If
one of my severs is blocking offending traffic, I'd like all of them to go
ahead and block it. I've done something like this by using fail2ban to post
(via HTTP) the attacker information to another server.. That server uses
fail2ban to scrape the http logs and blocks the offending traffic as well..
Works well for hub and spoke, but not for mesh setups.

-Brett
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Presence server performance

2010-11-15 Thread John Khvatov 
Hello all.

I'm testing performance of OpenSIPS presence server. I got bad results... Any 
ideas how to improve it?

My test scenario (SIPp xml config: http://dev.sgu.ru/pub/pubsub.xml):

SIPpPresence Server
 | PUBLISH |
 |>|
 | |
 | 200 OK  |
 |<|
 | |
 | SUBSCRIBE (on publised uri) |
 |>|
 | |
 | 200 OK  |
 |<|
 | |
 | NOTIFY  |
 |<|
 | |
 | 200 OK  |
 |>|

After 10.000 calls (10.000 scenario) OpenSIPS began to write warnings about 
memory to log:
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation

Results with 100 calls per second:
Total calls: 74454
Calls per second: 16.485 cps
Successful calls:  73772
Failed calls: 382
(With 10.000 "total calls" OpenSIPS shows 100 cps, almost zero failed calls)


Results with 1000 calls per second:
Total calls 20857
Calls per second: 12.696 cps
Successful calls: 15360
Failed calls: 2497

SUBSCRIBE to one of 500.000 published user is running 1-2 sec..

System configuration.
OS: Debian squeeze/sid (Not tuned)
Processor: Intel Xeon E5345
RAM: 1G (OpenSIPS executed with -m 64 option. Memory is still available during 
tests)
OpenSIPS: Version 1.6.3. Config is very simple: handle_subscribe() && 
handle_publish(). fallback2db disabled. Storage postgresql 8.4.

-- 
WBR, John Khvatov


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] b2bua core dump & db truncate

2010-11-15 Thread Anca Vamanu 

Hi,

Have you by any chance loaded the b2b_logic module before b2b_entities 
module?


Regards,
Anca

On 11/14/2010 12:18 AM, thrillerbee wrote:

New core dump on rev 7371
Backtrace attached.

Thanks.

On Thu, Nov 11, 2010 at 4:41 AM, Anca Vamanu > wrote:


Hi,

You were right, sorry, I did a partial commit. It is now ok.

Regards,
Anca



-- 
Anca Vamanu

www.voice-system.ro 


___
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
   



--
Anca Vamanu
www.voice-system.ro

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] b2bua core dump & db truncate

2010-11-15 Thread thrillerbee 
Actually, yes, I have.  Is that incorrect?  I'll reverse the order they're
loaded & test.

Thanks.

On Mon, Nov 15, 2010 at 11:37 AM, Anca Vamanu  wrote:

>  Hi,
>
> Have you by any chance loaded the b2b_logic module before b2b_entities
> module?
>
> Regards,
> Anca
>
>
> On 11/14/2010 12:18 AM, thrillerbee wrote:
>
> New core dump on rev 7371
> Backtrace attached.
>
>  Thanks.
>
> On Thu, Nov 11, 2010 at 4:41 AM, Anca Vamanu  wrote:
>
>> Hi,
>>
>> You were right, sorry, I did a partial commit. It is now ok.
>>
>> Regards,
>> Anca
>>
>>
>>
>> --
>> Anca Vamanu
>> www.voice-system.ro
>>
>>
>>  ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> ___
> Users mailing 
> listus...@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Anca Vamanuwww.voice-system.ro
>
>
> ___
> 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

Re: [OpenSIPS-Users] b2bua core dump & db truncate

2010-11-15 Thread Anca Vamanu 
They must be loaded in reverse order because b2b_logic depends on 
b2b_entities  - but it shouldn't crash, I will fix that.


Thanks also.

On 11/15/2010 07:41 PM, thrillerbee wrote:
Actually, yes, I have.  Is that incorrect?  I'll reverse the order 
they're loaded & test.


Thanks.

On Mon, Nov 15, 2010 at 11:37 AM, Anca Vamanu > wrote:


Hi,

Have you by any chance loaded the b2b_logic module before
b2b_entities module?

Regards,
Anca


On 11/14/2010 12:18 AM, thrillerbee wrote:

New core dump on rev 7371
Backtrace attached.

Thanks.

On Thu, Nov 11, 2010 at 4:41 AM, Anca Vamanu mailto:a...@opensips.org>> wrote:

Hi,

You were right, sorry, I did a partial commit. It is now ok.

Regards,
Anca



-- 
Anca Vamanu

www.voice-system.ro 


___
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
   



-- 
Anca Vamanu

www.voice-system.ro  
 



___
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
   



--
Anca Vamanu
www.voice-system.ro

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] b2bua core dump & db truncate

2010-11-15 Thread thrillerbee 
Anca,

I upgraded to rev 7383 and changed config to:

...
loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
...

It crashed after about 15 minutes of moderate traffic.  It actually wrote 2
core files so I've attached both backtraces (first is *_e.txt, second is
*_2.txt).

Thanks again.

On Mon, Nov 15, 2010 at 11:44 AM, Anca Vamanu  wrote:

>  They must be loaded in reverse order because b2b_logic depends on
> b2b_entities  - but it shouldn't crash, I will fix that.
>
> Thanks also.
>
>
> On 11/15/2010 07:41 PM, thrillerbee wrote:
>
> Actually, yes, I have.  Is that incorrect?  I'll reverse the order they're
> loaded & test.
>
>  Thanks.
>
> On Mon, Nov 15, 2010 at 11:37 AM, Anca Vamanu  wrote:
>
>>  Hi,
>>
>> Have you by any chance loaded the b2b_logic module before b2b_entities
>> module?
>>
>> Regards,
>> Anca
>>
>>
>> On 11/14/2010 12:18 AM, thrillerbee wrote:
>>
>> New core dump on rev 7371
>> Backtrace attached.
>>
>>  Thanks.
>>
>> On Thu, Nov 11, 2010 at 4:41 AM, Anca Vamanu  wrote:
>>
>>> Hi,
>>>
>>> You were right, sorry, I did a partial commit. It is now ok.
>>>
>>> Regards,
>>> Anca
>>>
>>>
>>>
>>> --
>>> Anca Vamanu
>>> www.voice-system.ro
>>>
>>>
>>>  ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> ___
>> Users mailing 
>> listus...@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> --
>> Anca Vamanuwww.voice-system.ro
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> ___
> Users mailing 
> listus...@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> Anca Vamanuwww.voice-system.ro
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
Core was generated by `/usr/local/sbin/opensips -P 
/var/run/opensips/opensips.pid -m 2048 -u root -g r'.
Program terminated with signal 11, Segmentation fault.
[New process 23158]
#0  0xb72f5a1c in t_uac (method=0xb7277498, headers=0x0, body=0x0, 
dialog=0x81a5b90, cb=0, cbp=0x0, release_func=0) at ../../mem/../hash_func.h:56
56  v=(*p<<24)+(p[1]<<16)+(p[2]<<8)+p[3];
(gdb) bt
#0  0xb72f5a1c in t_uac (method=0xb7277498, headers=0x0, body=0x0, 
dialog=0x81a5b90, cb=0, cbp=0x0, release_func=0) at ../../mem/../hash_func.h:56
#1  0xb72f72c3 in req_within (method=0xb7277498, headers=0x0, body=0x0, 
dialog=0x81a5b90, completion_cb=0, cbp=0x0, release_func=0) at uac.c:396
#2  0xb726a340 in b2b_send_req (dlg=0x3985d8dc, leg=0x81a576c, 
method=0xb7277498, extra_headers=0x0) at dlg.c:1701
#3  0xb72713b6 in b2b_tm_cback (t=0x3c4db4f0, htable=0x37276efc, ps=0xb73018d4) 
at dlg.c:1983
#4  0xb7264fd2 in b2b_client_tm_cback (t=0x3c4db4f0, type=512, ps=0xb73018d4) 
at client.c:44
#5  0xb72dc924 in run_trans_callbacks (type=512, trans=0x3c4db4f0, req=0x0, 
rpl=0x81a5088, code=200) at t_hooks.c:208
#6  0xb72f36f6 in local_reply (t=0x3c4db4f0, p_msg=0x81a5088, branch=0, 
msg_status=200, cancel_bitmap=0xbfc09c80) at t_reply.c:1348
#7  0xb72f4a39 in reply_received (p_msg=0x81a5088) at t_reply.c:1493
#8  0x080678c2 in forward_reply (msg=0x81a5088) at forward.c:559
#9  0x080986bb in receive_msg (
buf=0x81783a0 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 
207.36.90.52;branch=z9hG4bK7bad.deb109c1.0\r\nContact: 
\r\nTo: 
;tag=58dcb4ae-co6123-INS001\r\nFrom: , u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 
0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}
new_cell = (struct cell *) 0x3d2473a0
backup = 
buf = 
buf1 = 
buf_len = 
buf_len1 = 
ret = 
flags = 
backup_route_type = 
hi = 
send_sock = 
req = (struct sip_msg *) 0x0
__FUNCTION__ = "t_uac"
#1  0xb72f72c3 in req_within (method=0xb7277498, headers=0x0, body=0x0, 
dialog=0x81a5b90, completion_cb=0, cbp=0x0, release_func=0) at uac.c:396
__FUNCTION__ = "req_within"
#2  0xb726a340 in b2b_send_req (dlg=0x3985d8dc, leg=0x81a576c, 
method=0xb7277498, extra_headers=0x0) at dlg.c:1701
td = (dlg_t *) 0x0
result = 
__FUNCTION__ = "b2b_send_req"
#3  0xb72713b6 in b2b_tm_cback (t=0x3c4db4f0, htable=0x37276efc, ps=0xb73018d4) 
at dlg.c:1983
hash_index = 3090
local_index = 3665002
b2b_cback = (b2b_notify_t) 0xb7255efb 
dlg = (b2b_dlg_t *) 0x3985d8dc
param = {s = 0x81b21ac "3090.8", len = 6}
statuscode = 200
leg = (dlg_leg_t *) 0x81a576c
pto = 
TO = {error = 135933872, body = {s = 0x81a5088 "/\034", len = 0}, uri = 
{s = 0xb72bb73d "\205##017\210\234\002", len = -1077896652}, display = {
s = 0x81a5524 "D\204\027\b\026

[OpenSIPS-Users] opensips BYE issue

2010-11-15 Thread Tomasz 
Hello,

Can you helo me with some issue?
I have such scenario:

Dialer registered to asterisk via outbound proxy (TCP) and XLite connected
to the same asterisk via UDP.
When I make a call from dialer to XLite I have no problem until I want to
end a call on XLite side.

When XLite disconnects a call, than on dialer side the call is not finished.
Wireshark logs show all comunication goes via TCP but BYE is sent from
opensips via UDP.

Is it possible to force opensips to send BYE via TCP too.
I tried:

if (is_method("BYE") && src_ip=="xx.xx.xx.xx")
{
search_append('Request-URI:.*sip:[^>[:cntrl:]]*', ';transport=tcp');
xlog("L_INFO", " $ru \n");
}

but this looks like not working for me.
RURI port and IP is correct but transport is not set to TCP.

Can you help me?
\
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] (CDRTool) combining sip_trace/media records from multiple proxies

2010-11-15 Thread Jeff Pyle 
Great!

Another question:  As a packet passes from proxy A to proxy B, won't both 
proxies write basically the same packet to the sip_trace table?  In this 
instance CDRTool will simply display both packets, no?


- Jeff


From: Adrian Georgescu mailto:a...@ag-projects.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Mon, 15 Nov 2010 03:42:17 -0500
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] (CDRTool) combining sip_trace/media records from 
multiple proxies

Yes you can, just define the data sources for sip trace and media trace to 
point to the same databases.

Adrian

On Nov 11, 2010, at 12:38 PM, Jeff Pyle wrote:

Hello,

In my network I have three Opensips instances routing traffic to, from and 
across my network.  I use MySQL with radius for Opensips accounting, as well as 
MySQL for sip_trace and MySQL + radius for Mediaproxy accouting.  I use CDRTool 
to view the records together.

In CDRTool I have a separate data source for each proxy's records, as each is 
written to a separate database.  That's fine.  This also leads to separate 
sip_trace and mediaproxy records for the same call ID.

1)  Is it possible to use the same sip_trace table for all three proxies, and 
therefore view the traffic across all three proxies at the same time in CDRTool 
no matter which proxy's traffic I choose to look at?  My hope here is they will 
key on the same call ID.  All three proxies are VMs on the same dom0 so their 
clocks are the same.

2)  Mediaproxy is engaged only once across any of the three proxies for the 
same call.  Can I use the same media_sessions table for all three dispatchers 
to see the media records for a call no matter which proxy's traffic I'm looking 
at?  The MySQL side seems easier here than the radius one.  I fear at best I'd 
lose the view of the radius data unless I pick the "correct" proxy in CDRTool.

My goal is to minimize the amount of places I have to look when looking for the 
same data.

Any thoughts?  Am I trying to combine apples and pomegranates?


- Jeff

___
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

Re: [OpenSIPS-Users] Dispatcher radius messages are not valid

2010-11-15 Thread Bogdan-Andrei Iancu 

Hi Maciej,

Sounds quite clear (from the err message) that the secrets on radius 
server and radius client are not the sameIt is not an opensips 
issue, it is a matter of configuring the radius server and radius client 
library.


Regards,
Bogdan

Maciej Bylica wrote:

Hello,

I am working on opensips 1.6.3 $Revision: 4448 together with
media-dispatcher 2.4.3, media-relay 2.4.3, python 2.5.2-3, freeradius
2.1.8, radiusclient-ng 0.5.6
Freeradius should gather radius messages directly from opensips and
dispatcher. Both are installed on the same server and use the same
radiusclient.conf file.

The problem is that radius messages generated from dispatcher are not
taken into account while i have no problem with opensips radius
messages (secred for dispatcher and opensips is the same)
Here is an output from radius server

Waking up in 0.10 seconds.
Thread 9 got semaphore
Thread 9 handling request 121, (13 handled so far)
[] Received Accounting-Request packet from client 10.1.1.229
with invalid signature!  (Shared secret is incorrect.) Dropping packet
without response.

I've already tested freeradius-xs from debian pkg with same effect.
I am running 32bit os linux debian lenny.

Has anybody similiar problem. Could you guys pls point me what should i check?

Thx in advance,
Maciej

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

  



--
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] BLA/SLA not working

2010-11-15 Thread duane . larson 
I see. I will post on Snom Forum and maybe they can get this corrected at a  
later date.


Thanks for looking at it!



On Nov 11, 2010 9:08am, Anca Vamanu  wrote:

Hi Duane,






 From the logs it seems that Snom does not respect the format of  
the "dialog-info" document and sometimes it does not include  
a "direction" in the dialog node, this is considered compulsory by  
OpenSIPS to match the dialog.







Regards,







--




Anca Vamanu




www.voice-system.ro













On 11/10/2010 11:39 PM, duane.lar...@gmail.com wrote:





I am following the Tutorial to configure BLA and can't get it to work. I  
believe my issue has to do with my domain parts in the NOTIFY and PUBLISH  
messages. When two Snom phones have the same line configured for a button  
I am able to call the BLA DID have have them both ring. When one phone  
picks up the line the other phone is told the call has been picked up and  
stops ringing. The only thing that isn't working is that the line light  
is not lighting up to show that someone is on the line. Same thing when  
the user puts the call on hold. From an NGREP perspective I see that  
NOTIFY's are sent to OpenSIPS, but OpenSIPS doesn't send any notifies to  
the other BLA users to tell them the line is in use. In my OpenSIPS  
syslogs I see a bunch of Presence:error's






When the phones register or send Notifies/Publishes does the domain part  
need to be the domain like xyz.com or should it be the IP that shows up  
in location table?







Can someone tell me what I am doing wrong.







Here is the OpenSIPS config









___




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

[OpenSIPS-Users] radius debugging

2010-11-15 Thread Brett Nemeroff 
Hello List,
I'm wondering if there is a good way to debug radius requests. I'm sending
Auth requests via radius_send_auth. I'm pretty sure my sets are broken, but
I can't really see what it's attempting to send out. The radius server is on
another server and isn't a local freeradius.

One of my issues is that I need to send a param like:
Cisco-AVPair="h323-gw-address=1.2.3.4"

I'm not sure how to populate that into a set. I've tried to set up the
whole "h323-gw-address=1.2.3.4" bit as the contents of an avp like
$avp(s:some_avp)="h323-gw-address=1.2.3.4";

and using that AVP as the value in the set. But that doesn't work either.

Any advice?

Thanks,
Brett
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] acconting messages

2010-11-15 Thread Denis Putyato 
Hello!

 

Is there any chancy for accounting calls which were finished by sending failure 
code using send_reply() func.?

 

Thank you

 

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] acconting messages

2010-11-15 Thread Ovidiu Sas 
You can do manual accounting:
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id294003

Or, you can create a new transaction, flag it for acc and then
terminate it t_reply:
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293687


Regards,
Ovidiu Sas

On Tue, Nov 16, 2010 at 12:30 AM, Denis Putyato  wrote:
> Hello!
>
> Is there any chancy for accounting calls which were finished by sending
> failure code using send_reply() func.?
>
> Thank you

___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] acconting messages

2010-11-15 Thread Denis Putyato 
Thank you for reply

First variant is not quite flexible for me.
The second variant more interesting, but it doesn't work

A piece of code from opensips.cfg:

modparam("tm", "fr_timer", 10)
modparam("tm", "wt_timer", 30)
modparam("tm", "fr_inv_timer_avp", "$avp(i:25)")
modparam("tm", "T1_timer", 1000)
...
modparam("acc", "db_flag", 15)
modparam("acc", "db_missed_flag", 16)
modparam("acc", "failed_transaction_flag", 17)
modparam("acc", "db_table_acc", "acc")
modparam("acc", "db_table_missed_calls", "acc")
...

 if ($avp(i:200)==1) {
 t_newtran();
 setflag(16);
 setflag(17);
 t_flush_flags();
 t_reply("403", "Forbidden_gw");
 exit;
 }

And after this there is no records in ACC table.
May be I do something wrong?


-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: Tuesday, November 16, 2010 8:40 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] acconting messages

You can do manual accounting:
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id294003

Or, you can create a new transaction, flag it for acc and then
terminate it t_reply:
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293687


Regards,
Ovidiu Sas

On Tue, Nov 16, 2010 at 12:30 AM, Denis Putyato  wrote:
> Hello!
>
> Is there any chancy for accounting calls which were finished by sending
> failure code using send_reply() func.?
>
> Thank you

___
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