Re: [OpenSIPS-Users] Help me on openxcap

[Adrian Georgescu]

Mahesh,

You must read RFC 4825 for this

http://www.tools.ietf.org/html/rfc4825

Adrian

On Nov 13, 2008, at 8:24 AM, <[EMAIL PROTECTED]> <[EMAIL PROTECTED] 
 > wrote:

> Hi All,
>
> I am working on openxcap 1.0.6 version.
> It is running fine. But when i am trying to upload an xml file with  
> UCTIMS client, i specified the URI of xcap as  
> http://xcap.info-spectrum.com/xcap-root 
> .
> Its giving invalid URI.
> My question is : "How to define a valid URI? Please Help me on this ".
> Regards,
> Mahesh
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Re: [OpenSIPS-Users] CDRTool problem

[Adrian Georgescu]

Hi Dilip,

Did you read the RATING.txt document?

http://cdrtool.ag-projects.com/browser/doc/RATING.txt

Adrian


On Nov 13, 2008, at 2:27 PM, Dilip wrote:


Hello Evrybody,
I am using CDRTool  for the accounting.It GUI is working fine.
But i wants to add the rating in its GUI.For that i have read the  
Rating

Engine Documentation but its not affected.
Whats the exact procedure to display the call rate ??
which r the tables require to edit ???
Can anybody help me for that.

Regards,
Dilip Modi

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Re: [OpenSIPS-Users] Error relaying ACK to the same IP but different port

[Adrian Georgescu]

Gustavo,

If you post your configuration you score more chances to get useful  
help from somebody. What you have just showed can have an infinite  
number of reasons without a configuration.


Adrian

On Nov 13, 2008, at 6:02 PM, Gustavo Mistrinelli wrote:

Hello, I have a schema composed by OpenSIPS (10.20.200.1:5060) and  
B2BUA-Asterisk (10.20.200.1:5070) ,


Here goes a call flow from a hardphone (10.10.115.113) to a  
Conference Number located in another server (Asterisk: 192.168.0.1)
I found an issue, see timeframe 0.028, you can see OpenSIPS sending  
an ACK to itself at the same port(5060) when it should be to 5070,  
Then B2BUA send again 200OK to OpenSIPS..and go on...
If I put B2BUA in another server using and same port (5070) it works  
fine.. :S


|Time | 10.10.115.113 | 10.20.200.1   |
|0.000| INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] 
 To:sip:[EMAIL PROTECTED]

| |(5060)   -->  (5060)   |
|0.000| 407 Proxy Authentication Required |SIP Status
| |(5060)   <--  (5060)   |
|0.004| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|0.009| INVITE SDP ( telephone-event) |SIP From: sip:[EMAIL PROTECTED] 
 To:sip:[EMAIL PROTECTED]

| |(5060)   -->  (5060)   |
|0.009| 100 Trying|   |SIP Status
| |(5060)   <--  (5060)   |
|0.012|   | INVITE SDP ( telephone- 
event)SIP Request

| |   |(5060)   -->(5070)
|0.012|   | 100 Trying|SIP Status
| |   |(5070)   -->(5060)
|0.013|   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)
|0.013| 200 OK SDP ( telephone-event) |SIP Status
| |(5060)   <--  (5060)   |
|0.027| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|0.028|   | ACK   |SIP Request
| |   |(5060)   -->(5060)
|1.021|   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)
|1.021| 200 OK SDP ( telephone-event) |SIP Status
| |(5060)   <--  (5060)   |
|1.038| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|1.038|   | ACK   |SIP Request
| |   |(5060)   -->(5060)
|2.021|   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)
|2.021| 200 OK SDP ( telephone-event) |SIP Status
| |(5060)   <--  (5060)   |
|2.036| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|2.036|   | ACK   |SIP Request
| |   |(5060)   -->(5060)
|4.021|   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)
|4.021| 200 OK SDP ( telephone-event) |SIP Status
| |(5060)   <--  (5060)   |
|4.032| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|4.032|   | ACK   |SIP Request
| |   |(5060)   -->(5060)
|8.021|   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)
|8.021| 200 OK SDP ( telephone-event) |SIP Status
| |(5060)   <--  (5060)   |
|8.033| ACK   |   |SIP Request
| |(5060)   -->  (5060)   |
|8.034|   | ACK   |SIP Request
| |   |(5060)   -->(5060)
|12.021   |   | 200 OK SDP ( telephone- 
event)SIP Status

| |   |(5070)   -->(5060)

Thanks in advance,
--
Gustavo Mistrinelli
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Re: [OpenSIPS-Users] [OT] about scalability, experiences, demands

[Adrian Georgescu]

From a customer point of view is great that you are satisfied with  
the software.


Myself as a platform vendor having to satisfy the needs of multiple  
customers I can only concur with Bogdan that the curent design has  
flaws inherited from the original requirements and fixing them one by  
one by developing or improving modules to navigate around them is not  
the most efficient way forward.


A consistent core with a generic API to a higher level application  
that does not depend much on the core version and where a programming  
language chosen by the customer can be used is much more future proof  
then patching endlessly the existing code with lose modules and having  
to rewrite the configuration with every major version upgrade.


Having two projects and two ways to achieve the same goals may help  
the customers in the future.


Adrian

On Nov 14, 2008, at 2:59 PM, Henning Westerholt wrote:


Hello all,

recently some statement came to my attention that "there is a common  
consent
that the current design/architecture of [..] OpenSER (inherited from  
SER) is
no longer able to deliver and to meet the present requirements and  
demands".


I don't want to argue that much about this opinion, in fact the  
demands to a
Voice over IP solution depends very much on the certain setup. But i  
want to
share some details from my experiences in developing and operating a  
big VoIP

infrastructure here at 1&1.

We've about 2 million customers on our platform, that uses over 5  
million
individual numbers and terminate about 1 billion minutes per month.  
We're

able to provide a good service with the actual architecture of OpenSER
without any real problems. Of course there is always some room for
improvements, but so far the main challenges we faced were not in the
scalability or performance areas. More important issues are e.g. the  
inherent

complexity of the SIP protocol and the maintainance of a good quality
assurance and integration process.

We started some years ago with OpenSER 0.9.5, which we then extended  
a lot in

house. For example we implemented more than 25 own modules, a own path
implementation, did a lot of bug fixing and workarounds for certain  
problems
we've found. We're able to reduce this amount of proprietary code a  
lot in
the past, because of progress in the OpenSER development,  
intregration of
our "key" modules and a lot of other contributions. We're using now  
something
between OpenSER 1.3 and Kamailio 1.4 with only a few private  
extensions.


So in my opinion the actual design of our server is not "[..] an  
inevitable
dead-end that needs to be avoided.". I rather think that we'll be  
able with
continuing improvements to tackle the upcoming challenges well,  
especially as
we will work together in the future with the SER developers in  
improving

important areas of this software.

But this is just my personal opinion, everybody is of course free to  
have

their own position.

With best regards,

Henning Westerholt

--
Henning Westerholt - Development Consumer Products / DSL Core
1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

Vorstände: Henning Ahlert, Ralph Dommermuth, Matthias Ehrlich, Thomas
Gottschlich, Robert Hoffmann, Markus Huhn, Hans-Henning Kettler,
Dr. Oliver Mauss, Jan Oetjen - Aufsichtsratsvorsitzender: Michael  
Scheeren

Amtsgericht Montabaur / HRB 6484

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Re: [OpenSIPS-Users] search problem with CDRTool

[Adrian Georgescu]

Add the field quota to the MySQL subscriber table.

Adrian


On Nov 17, 2008, at 4:48 AM, troxlinux wrote:

Hi list , does a few days installs CDRTool in my server  openser, I  
am using radius and to openser to handle accounting,  but I have a  
problem that shows me every time that I search for the calls made by  
my users from a date to another.


11 records found. Database error: Invalid SQL: select quota from  
subscriber where username = '112' and domain = '192.168.10.1' MySQL  
error: 1054 (Unknown column 'quota' in 'field list') 59Session halted


Database error: Invalid SQL: select quota from subscriber where  
username = '119' and domain = '192.168.10.1' MySQL error: 1054  
(Unknown column 'quota' in 'field list') 59Session halted.


23 records found.
Database error: Invalid SQL: select quota from subscriber where  
username = 'asterisk' and domain = '192.168.10.1' MySQL error: 1054  
(Unknown column 'quota' in 'field list') 59Session halted.


any idea that it can be failing?

best regards

rickygm
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Re: [OpenSIPS-Users] CDRTool Not Showing the Price

[Adrian Georgescu]

Hi Dilip,

I canot see any calls at all in your syslog entries. As a wild guess  
most probably your destination field is not formatted correctly, it  
should start with 00 to be matched by the standard destination lookup  
logic.

Can you paste the Radius accounting tickets for one  OpenSER session  
(START, STOP, UPDATE radius tickets)?

You can find the radius tickets under /var/log/freeradius/radacct/*

Regards,
Adrian


On Nov 17, 2008, at 1:25 PM, Dilip wrote:

> hello Sir,
> Actually i could not see the price in the CDRTool GUI.
> I have configured it according to the rating document.but Still  
> there is
> not any calculation is start by cdrtool.
> What may be the problem ???
> I have paste here the syslog message...so please guide me.


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Re: [OpenSIPS-Users] CDRTool Not Showing the Price

[Adrian Georgescu]

Have you read the RATING.txt document?

3. Determination of the destination id

This section is relevant and you have not done any Radius setup to  
store the CanonicalURI  in your openser and mediaproxy as far as I can  
see from your examples:


Adrian


On Nov 18, 2008, at 9:43 AM, Dilip wrote:


Hi Adrian,
I have paste the radacct ticket below.


Mon Nov 10 03:14:46 2008
 Acct-Status-Type = Start
 Service-Type = Sip-Session
 Sip-Response-Code = 200
 Sip-Method = INVITE
 Event-Timestamp = "Nov 10 2008 03:14:46 EST"
 Sip-From-Tag = "e4714a611fffea51"
 Sip-To-Tag = "10114108130863030106211833"
 Acct-Session-Id = "[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>"
 Called-Station-Id = "sip:[EMAIL PROTECTED]:3300;user=phone 
"

 Calling-Station-Id = "sip:[EMAIL PROTECTED]:3300;user=phone"
 X-Ascend-IPX-Peer-Mode =
0x7369703a3133303337323530304036372e3230352e38352e3131393a30303b757365723d70686f6e65
 User-Name = "10009"
 Attr-111 = 0x36372e3230352e38352e313139
 X-Ascend-Send-Secret = 0x35382e36382e3132332e3531
 X-Ascend-Receive-Secret = 0x3634353235
 X-Ascend-FR-Direct-DLCI =
0x2275726d6922203c7369703a31303030394036372e3230352e38352e3131393a30303b757365723d70686f6e653e3b7461673d6534373134613631316665613531
 X-Ascend-Handle-IPX =
0x4772616e6473747265616d20425431323020312e312e302e32
 NAS-IP-Address = 192.168.1.56
 NAS-Port = 5060
 Acct-Delay-Time = 0
 Client-IP-Address = 192.168.1.56
 Acct-Unique-Session-Id = "4a99b229fe5c3e83"
 Timestamp = 1226304886

Mon Nov 10 03:14:46 2008
 Acct-Status-Type = Stop
 Service-Type = Sip-Session
 Sip-Response-Code = 200
 Sip-Method = BYE
 Event-Timestamp = "Nov 10 2008 03:14:46 EST"
 Sip-From-Tag = "e4714a611fffea51"
 Sip-To-Tag = "10114108130863030106211833"
 Acct-Session-Id = "[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>"
 Called-Station-Id = "sip: 
[EMAIL PROTECTED]:3300;user=phone"

 Calling-Station-Id = "sip:[EMAIL PROTECTED]:3300;user=phone"
 X-Ascend-IPX-Peer-Mode =
0x7369703a3133303337323530304036372e3230352e38352e3131393a30303b757365723d70686f6e65
 User-Name = "10009"
 Attr-111 = 0x36372e3230352e38352e313139
 X-Ascend-Send-Secret = 0x35382e36382e3132332e3531
 X-Ascend-Receive-Secret = 0x3634353235
 X-Ascend-FR-Direct-DLCI =
0x2275726d6922203c7369703a31303030394036372e3230352e38352e3131393a30303b757365723d70686f6e653e3b7461673d6534373134613631316665613531
 X-Ascend-Handle-IPX =
0x4772616e6473747265616d20425431323020312e312e302e32
 NAS-IP-Address = 192.168.1.56
 NAS-Port = 5060
 Acct-Delay-Time = 0
 Client-IP-Address = 192.168.1.56
 Acct-Unique-Session-Id = "4a99b229fe5c3e83"
 Timestamp = 1226304886


Please give me any idea so that i can complete it.

Regards,
Dilip

Adrian Georgescu wrote:

Hi Dilip,

I canot see any calls at all in your syslog entries. As a wild guess
most probably your destination field is not formatted correctly, it
should start with 00 to be matched by the standard destination lookup
logic.

Can you paste the Radius accounting tickets for one  OpenSER session
(START, STOP, UPDATE radius tickets)?

You can find the radius tickets under /var/log/freeradius/radacct/*

Regards,
Adrian


On Nov 17, 2008, at 1:25 PM, Dilip wrote:


hello Sir,
Actually i could not see the price in the CDRTool GUI.
I have configured it according to the rating document.but Still  
there is

not any calculation is start by cdrtool.
What may be the problem ???
I have paste here the syslog message...so please guide me.






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Re: [OpenSIPS-Users] CDRTool Not Showing the Price

[Adrian Georgescu]

Hi Dilip,

I canot see any calls at all in your syslog entries. As a wild guess  
most probably your destination field is not formatted correctly, it  
should start with 00 to be matched by the standard destination lookup  
logic.


Can you paste the Radius accounting tickets for one  OpenSER session  
(START, STOP, UPDATE radius tickets)?


You can find the radius tickets under /var/log/freeradius/radacct/*

Regards,
Adrian


On Nov 17, 2008, at 1:25 PM, Dilip wrote:


hello Sir,
Actually i could not see the price in the CDRTool GUI.
I have configured it according to the rating document.but Still  
there is

not any calculation is start by cdrtool.
What may be the problem ???
I have paste here the syslog message...so please guide me.

Nov 17 06:50:01 cl-t041-080cl cdrtool[29397]: Unlock
asterisk_vm:asterisk_cdr
Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Normalize datasource
ser_radius, database DB_CDRTool, table radacct
Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Normalize lock id 12325
aquired for ser_radius:radacct
Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Normalize datasource
asterisk_vm, database DB_radius, table asterisk_cdr
Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Normalize lock id 12325
aquired for asterisk_vm:asterisk_cdr
Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Unlock  
ser_radius:radacct

Nov 17 06:55:01 cl-t041-080cl cdrtool[29434]: Unlock
asterisk_vm:asterisk_cdr
Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Normalize datasource
ser_radius, database DB_CDRTool, table radacct
Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Normalize lock id 12344
aquired for ser_radius:radacct
Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Normalize datasource
asterisk_vm, database DB_radius, table asterisk_cdr
Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Normalize lock id 12344
aquired for asterisk_vm:asterisk_cdr
Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Unlock  
ser_radius:radacct

Nov 17 07:00:01 cl-t041-080cl cdrtool[29471]: Unlock
asterisk_vm:asterisk_cdr
Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Normalize datasource
ser_radius, database DB_CDRTool, table radacct
Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Normalize lock id 12348
aquired for ser_radius:radacct
Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Normalize datasource
asterisk_vm, database DB_radius, table asterisk_cdr
Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Normalize lock id 12348
aquired for asterisk_vm:asterisk_cdr
Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Unlock  
ser_radius:radacct

Nov 17 07:05:02 cl-t041-080cl cdrtool[29509]: Unlock
asterisk_vm:asterisk_cdr
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Normalize lock id 11823
aquired for ser_radius:radacct
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Memory usage: 8.52MB,
memory limit: 16MB
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Loaded 3 profiles
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Loaded 0 ratesHistory
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Loaded 0 holidays
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Loaded 1 enumTlds
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Memory usage: 8.53MB,
memory limit: 16MB
Nov 17 07:08:39 cl-t041-080cl cdrtool[19924]: Normalization done in  
0 s,

memory usage: 8.55 MB
Nov 17 07:08:40 cl-t041-080cl cdrtool[19924]: Unlock  
ser_radius:radacct


Please help me.
How to show the price in the CDRTool GUI.
I have import all the rating csv in the db.
below is my global.inc file of cdrtool.
Is there any mistake in the file.

#
# 1. Change all hostnames and passwords according to the installation
# 2. Copy this file to /etc/cdrtool/global.inc
#

###
# System and web paths

$CDRTool['tld']= "/CDRTool";
$CDRTool['Path']   = "/var/www/CDRTool";
$_PHPLIB['libdir'] = $CDRTool['Path']. "/phplib/";
include($_PHPLIB["libdir"] . "prepend.php3");

###
# PHP Error reporting
$errorReporting = (E_ALL & ~E_NOTICE);
$errorReporting = 1;// comment this out to enable PHP warnings
error_reporting($errorReporting);

###
# Service provider information

$CDRTool['provider']['name']  = "Provider name";
$CDRTool['provider']['service']   = "SIP service";
$CDRTool['provider']['timezone']  = "Europe/Amsterdam";
$CDRTool['provider']['fromEmail'] = "[EMAIL PROTECTED]";
$CDRTool['provider']['toEmail']   = "[EMAIL PROTECTED]";
$CDRTool['provider']['sampleLoginSubscriber'] = "[EMAIL PROTECTED]";
$CDRTool['provider']['sampleLoginDomain'] = "example.com";

###
# Rating engine settings
$RatingEngine=array("socketIP"   => "1.2.3.4",
   "socketPort" => "9024",
   "CDRS_class" => "ser_radius",
   "prepaid_lock"   => true,
   "log_delay"  => 0.05,
   "split_rating_table" 

Re: [OpenSIPS-Users] Help!! How to do failover of mysql connection

[Adrian Georgescu]

Try use this settings:

On master

auto_increment_increment = 2
auto_increment_offset= 0

On slave

auto_increment_increment = 2
auto_increment_offset= 1

This makes sure that auto-increments do not colide between master and  
slave so you can achieve switchover without conflicts because of auto  
increment columns.


Adrian


On Nov 26, 2008, at 10:18 PM, Brett Nemeroff wrote:

Before doing this, I'd seriously consider the problems associated  
with master-master replication.


um, I don't know what they are.. but I know they are real problems.  
Such as collisions in auto-incrementing data.

-Brett


On Wed, Nov 26, 2008 at 3:06 PM, Uwe Kastens <[EMAIL PROTECTED]> wrote:
Hi Krunal,

>
> I am just on my way to implement the same. At the moment I am  
planing a

> kind of Mysql Master - Master together with one VIP for
> Mysql-opensips-communication. So for your example you will  
have to

> mysql-servers. opensips is connection to lets say to 192.168.1.4
>  which
> points on one of your mysql-servers. You will need active- 
active for
> mysql since opensips will write some information in the  
database as

> well.
>
>
>
>  I am working with mysql-5 and heartbeat-2 in mode 1. Its nearly
> working.
>
>
> Would you please explain it somewhat in detail?

Sure, at the moment its only a testing platform. I took two
MySQL-Servers which are configured as master-master. So writing on  
both

databases is possible and is synchronized. I found a good step-by-step
guide under:

http://www.howtoforge.com/mysql_master_master_replication

I configured a very simple setup for heartbeat to share one VIP  
between
both servers for client connection. So a kind of mysql-test is  
needed to

check if the database is online on the "normal" IP-Adress. If not the
VIP should be switched to the other server. I found a kind of
nagios_mysql_check usefull.

The openser-servers are connecting only to the vip address. Read and
Write are working on only one database. This should be enough for a
small environment.

Why no master - slave? I found it to complicate to make a slave to a
master by skript.

Why not the "normal" setup (drbd with mysql and heartbeat)? I looks to
complicated.

But! I have no idea how it will work in a real world.

BR

kiste
--

kiste lat: 54.322684, lon: 10.13586

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Re: [OpenSIPS-Users] Mediaproxy or RtpProxy

[Adrian Georgescu]

Woody,

Statements that things behave poorly out of concrete context do not  
really help anyone.


The question is how many sessions to you want to handle? Can you  
answer this simple question? Then someone might be able to help you.


Adrian


On Nov 27, 2008, at 1:46 PM, Woody Dickson wrote:


Hi,

I am trying to decide on whether to use RtpProxy or MediaProxy for  
my current opensips implementation.  Previously MediaProxy was known  
to perform poorly under high loard.  Now with the latest 2.0  
release, does anyone know how does MediaProxy compare to RtpProxy in  
terms of performance, scalability, and ease-of-use?


I would appreciate any insight on this.

Thanks,
Woody
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Re: [OpenSIPS-Users] Mediaproxy or RtpProxy

[Adrian Georgescu]

For MediaProxy1 you need one machine for each 200 active sessions. For  
MediaProxy 2 you can use one machine only. I cannot speak for RTP  
proxy, you have to ask their developers


Adrian


On Nov 27, 2008, at 2:51 PM, Woody Dickson wrote:


Hi Adrian,

Sorry, I don't mean to offense anyone when I incorrectly used the  
term poorly.  Please accept my apology.


I would like to use either MediaProxy or RtpProxy to proxy 1-2K  
concurrent sessions.  Is this a realistic number for either one?


Could someone comment on the performance, scalability, and ease-of- 
use between the two?


Thanks,
Woody

On Thu, Nov 27, 2008 at 8:54 PM, Adrian Georgescu <[EMAIL PROTECTED] 
projects.com> wrote:

Woody,

Statements that things behave poorly out of concrete context do not  
really help anyone.


The question is how many sessions to you want to handle? Can you  
answer this simple question? Then someone might be able to help you.


Adrian


On Nov 27, 2008, at 1:46 PM, Woody Dickson wrote:


Hi,

I am trying to decide on whether to use RtpProxy or MediaProxy for  
my current opensips implementation.  Previously MediaProxy was  
known to perform poorly under high loard.  Now with the latest 2.0  
release, does anyone know how does MediaProxy compare to RtpProxy  
in terms of performance, scalability, and ease-of-use?


I would appreciate any insight on this.

Thanks,
Woody
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Re: [OpenSIPS-Users] Mediaproxy or RtpProxy

[Adrian Georgescu]

About easy of use, for MediaProxy2 you need to add a single line of  
configuration to opensips.cfg. Then there is a straight forward  
configuration file /etc/mediaproxy/config.ini with settings you might  
want to change if the defaults are not applicable for your setup.

Adrian

On Nov 27, 2008, at 2:51 PM, Woody Dickson wrote:

> Hi Adrian,
>
> Could someone comment on the performance, scalability, and ease-of- 
> use between the two?
>
> Thanks,
> Woody


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Re: [OpenSIPS-Users] Codec question

[Adrian Georgescu]

For the clarity codecs are only in the end-points (e.g. SIP phone).  
MediaProxy is not an end point and is codec agnostic. So it does not  
need to support a particular codec, it relays UDP/RTP packets  
regardless of what codec they have inside.


Adrian

On Nov 28, 2008, at 1:34 PM, Giuseppe Roberti wrote:


Hi.

OpenSIPS is not a media proxy.
It understand only SIP protocol.
You need some RTP proxy for dealing with audio/video (and so codecs).
The only two solution i know are:
 - rtpproxy http://rtpproxy.org/
 - mediaproxy http://mediaproxy-ng.org/

Regards.

Tseveendorj Ochirlantuu wrote:

Hello,
What kind of codecs included in OpenSIPS?  G.711mu, G711a ,G729 ...

Thank you

Sincerely,
Tseveen.





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[OpenSIPS-Users] SIP SIMPLE client release

[Adrian Georgescu]

Hello,

SIP SIMPLE client is a Python software library built on top of PJSIP  
that together with middleware allows for easy development of Client/ 
Server and Peer-to-Peer Internet communications end-points based on  
SIP SIMPLE protocols for voice, video, presence, instant messaging  
(IM) and file transfer capabilities.

The goal of this project is to deliver the best Open Source Python  
library for rich featured SIP end-points, while hiding the complex  
underlying functionality behind an easy to use high-level application- 
programming interface. This package supports to the SDP negotiation,  
audio codecs and NAT traversal functionality provided by PJSIP and  
will thus deliver rich communications combining instant messaging  
(IM), voice and video streams. It also supports file transfer and  
multi-party chat sessions using MSRP protocol, publication and  
subscription for rich presence information such as availability,  
moods, activities and geo-location, management for presence rules,  
resource lists, RLS services using XCAP protocol.

The software allows you to create elegant real-time communications  
applications without having to read the +1200 RFC documents behind it.

SIP SIMPLE client is closely developed together with and tested  
against the most popular SIP SIMPLE server software available today:  
OpenSIPS, OpenXCAP and MSRPRelay.

Included with the library, a set of command line tools are available  
for setting up audio, Instant Messaging and file transfer sessions,  
publish and subscribe to presence or other type of events.

• sip_register - REGISTER a SIP end-point with a SIP Registrar
• sip_audio_session - Setup a voice audio session (Voice over IP)
• sip_im_session - Setup IM session and File transfer using MSRP  
protocol
• sip_message - Send/receive text in page mode using SIP MESSAGE method
• sip_publish_presence - PUBLISH presence to a SIP Presence Agent
• sip_subscribe_presence - SUBSCRIBE to presence information
• sip_subscribe_winfo - SUBSCRIBE to watcher list on a SIP Presence  
Agent
• sip_subscribe_rls - SUBSCRIBE to lists managed by Resource List  
Server
• xcapclient - PUT/GET/DELETE full or partial documents on an XCAP  
server
• xcap_pres_rules - Manage content of pres-rules XCAP document
• xcap_rls_services - Manage content of RLS services XCAP document

Binary packages are available for Debian or Ubuntu on i386 and amd64  
architectures.

Installation instructions and the source code are available at:

http://SipSimpleClient.com

Feedback is welcome and if you wish to contribute or obtain support  
follow the information provided on the home page of the project.

Kind regards,
Adrian Georgescu


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Re: [OpenSIPS-Users] OpenXCAP and Open IMS

[Adrian Georgescu]

Try pres-rules instead of presence rules. If you PUT pres-rules you  
cannot obtain the same result back by doing GET presence-rules


Adrian


On Dec 5, 2008, at 5:39 PM, Kenneth Löfstrand wrote:


Hi
I have started playing around with the combination Open IMS/Presence,
OpenXCAP and OpenIMS. Currently OpenXCAP and the presence server are
installed in the same PC and the Open IMS stuff in other PC's. I can
have a terminal sending XCAP requests to OpenXCAP and I can subscribe
and publish to the presence server through the IMS network but the
presence server doesn't care about the presence rule documents.

When sending the XCAP PUT request I use the command:

PUT org.openmobilealliance.pres-rules/users/sip:[EMAIL PROTECTED]/pres-rules

no problem so far and the document can also be retreived.

When bob subscribes for alice presence information I can see that the
presence server sends a GET request using another AUID:

GET presence-rules/users/sip:[EMAIL PROTECTED]/index

and this results in a 404 error response. Why getting the index  
instead

of pres-rules?

From what I understand the AUID is defined in a different way by IETF
and OMA. How can this be handled by OpenIMS/OpenXCAP/OpenSIPS?

Best regards
Kenneth Löfstrand


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Re: [OpenSIPS-Users] OpenXCAP and Open IMS

[Adrian Georgescu]

It seems that on the README and wiki page the auid was incorrectly  
displayed. I have corrected it. You can use either these two auids to  
identify the presence rulles application:

- pres-rules
- org.openmobilealliance.pres-rules

Examples for how to PUT/GET/DELETE documents on the XCAP server are  
found here:

http://download.ag-projects.com/OpenXCAP/clients/README.xcapclient

You can also use the ready to use xcap_pres_rules.py script that does  
it interactively in the console:

http://sipsimpleclient.com/wiki/xcap_pres_rules

Regards,
Adrian

On Dec 6, 2008, at 11:51 AM, Kenneth Löfstrand wrote:

> So how do I try that? I cannot find where to specify which AUID to  
> be used by the presence server
>
> /Kenneth
>
>
> Adrian Georgescu skrev:
>> Try pres-rules instead of presence rules. If you PUT pres-rules you  
>> cannot obtain the same result back by doing GET presence-rules
>>
>>
>> Adrian
>>
>>
>>
>> On Dec 5, 2008, at 5:39 PM, Kenneth Löfstrand wrote:
>>
>>
>>> Hi
>>> I have started playing around with the combination Open IMS/ 
>>> Presence,
>>> OpenXCAP and OpenIMS. Currently OpenXCAP and the presence server are
>>> installed in the same PC and the Open IMS stuff in other PC's. I can
>>> have a terminal sending XCAP requests to OpenXCAP and I can  
>>> subscribe
>>> and publish to the presence server through the IMS network but the
>>> presence server doesn't care about the presence rule documents.
>>>
>>> When sending the XCAP PUT request I use the command:
>>>
>>> PUT org.openmobilealliance.pres-rules/users/sip:[EMAIL PROTECTED]/pres-rules
>>>
>>> no problem so far and the document can also be retreived.
>>>
>>> When bob subscribes for alice presence information I can see that  
>>> the
>>> presence server sends a GET request using another AUID:
>>>
>>> GET presence-rules/users/sip:[EMAIL PROTECTED]/index
>>>
>>> and this results in a 404 error response. Why getting the index  
>>> instead
>>> of pres-rules?
>>>
>>> From what I understand the AUID is defined in a different way by  
>>> IETF
>>> and OMA. How can this be handled by OpenIMS/OpenXCAP/OpenSIPS?
>>>
>>> Best regards
>>> Kenneth Löfstrand
>>>
>>>
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Re: [OpenSIPS-Users] error install python-xcaplib

[Adrian Georgescu]

Only Python 2.5 is supported (is mentioned in the README), and you are  
using Python 2.4


Adrian


On Dec 6, 2008, at 9:01 PM, troxlinux wrote:

Hi list ,  I am trying to install the  python-xcaplib to be able to  
install the server openxcap



python setup.py install
running install
running build
running build_py
running build_scripts
running install_lib
byte-compiling /usr/lib/python2.4/site-packages/xcaplib/ 
xcapclient.py to xcapclient.pyc
  File "/usr/lib/python2.4/site-packages/xcaplib/xcapclient.py",  
line 321

finally:
  ^
SyntaxError: invalid syntax
running install_scripts
changing mode of /usr/bin/xcapclient to 755

some idea because it fails when installing?

best regards

rickygm
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Re: [OpenSIPS-Users] OpenXCAP and Open IMS

[Adrian Georgescu]

May I ask which presence server uses presence-rules as AUID?

Adrian

On Dec 7, 2008, at 5:08 PM, Kenneth Löfstrand wrote:


Thanks for your replies, Adrian
to me OpenXCAP is generous enough to accept the two AUID's as you  
say but isn't the problem that I have to convince the presence  
server to use one of these. Currently it uses presence-rules as AUID.


Regards
Kenneth

Adrian Georgescu skrev:
It seems that on the README and wiki page the auid was incorrectly  
displayed. I have corrected it. You can use either these two auids  
to identify the presence rulles application:


- pres-rules
- org.openmobilealliance.pres-rules

Examples for how to PUT/GET/DELETE documents on the XCAP server are  
found here:


http://download.ag-projects.com/OpenXCAP/clients/README.xcapclient

You can also use the ready to use xcap_pres_rules.py script that  
does it interactively in the console:


http://sipsimpleclient.com/wiki/xcap_pres_rules

Regards,
Adrian

On Dec 6, 2008, at 11:51 AM, Kenneth Löfstrand wrote:

So how do I try that? I cannot find where to specify which AUID to  
be used by the presence server


/Kenneth


Adrian Georgescu skrev:
Try pres-rules instead of presence rules. If you PUT pres-rules  
you cannot obtain the same result back by doing GET presence-rules



Adrian



On Dec 5, 2008, at 5:39 PM, Kenneth Löfstrand wrote:



Hi
I have started playing around with the combination Open IMS/ 
Presence,
OpenXCAP and OpenIMS. Currently OpenXCAP and the presence server  
are
installed in the same PC and the Open IMS stuff in other PC's. I  
can
have a terminal sending XCAP requests to OpenXCAP and I can  
subscribe

and publish to the presence server through the IMS network but the
presence server doesn't care about the presence rule documents.

When sending the XCAP PUT request I use the command:

PUT org.openmobilealliance.pres-rules/users/sip:[EMAIL PROTECTED]/pres-rules

no problem so far and the document can also be retreived.

When bob subscribes for alice presence information I can see  
that the

presence server sends a GET request using another AUID:

GET presence-rules/users/sip:[EMAIL PROTECTED]/index

and this results in a 404 error response. Why getting the index  
instead

of pres-rules?

From what I understand the AUID is defined in a different way by  
IETF

and OMA. How can this be handled by OpenIMS/OpenXCAP/OpenSIPS?

Best regards
Kenneth Löfstrand


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Re: [OpenSIPS-Users] OpenXCAP and Open IMS

[Adrian Georgescu]

Can you try using the integrated mode where the server does not do  
XCAP queries but reads the documents directly from the database  
provisioned by the XCAP server.


The xcap_client module might use the wrong auid but I have never used  
it so I cannot advise what is wrong with your setup.


Adrian

On Dec 8, 2008, at 8:09 AM, Kenneth Löfstrand wrote:


I'm using the presence server that comes in Opensips 1.4

Regards
Kenneth


Adrian Georgescu skrev:

May I ask which presence server uses presence-rules as AUID?


Adrian


On Dec 7, 2008, at 5:08 PM, Kenneth Löfstrand wrote:



Thanks for your replies, Adrian
to me OpenXCAP is generous enough to accept the two AUID's as you  
say but isn't the problem that I have to convince the presence  
server to use one of these. Currently it uses presence-rules as  
AUID.


Regards
Kenneth

Adrian Georgescu skrev:

It seems that on the README and wiki page the auid was  
incorrectly displayed. I have corrected it. You can use either  
these two auids to identify the presence rulles application:


- pres-rules
- org.openmobilealliance.pres-rules

Examples for how to PUT/GET/DELETE documents on the XCAP server  
are found here:


http://download.ag-projects.com/OpenXCAP/clients/README.xcapclient

You can also use the ready to use xcap_pres_rules.py script that  
does it interactively in the console:


http://sipsimpleclient.com/wiki/xcap_pres_rules

Regards,
Adrian

On Dec 6, 2008, at 11:51 AM, Kenneth Löfstrand wrote:

So how do I try that? I cannot find where to specify which AUID  
to be used by the presence server


/Kenneth


Adrian Georgescu skrev:
Try pres-rules instead of presence rules. If you PUT pres-rules  
you cannot obtain the same result back by doing GET presence- 
rules



Adrian



On Dec 5, 2008, at 5:39 PM, Kenneth Löfstrand wrote:



Hi
I have started playing around with the combination Open IMS/ 
Presence,
OpenXCAP and OpenIMS. Currently OpenXCAP and the presence  
server are
installed in the same PC and the Open IMS stuff in other PC's.  
I can
have a terminal sending XCAP requests to OpenXCAP and I can  
subscribe
and publish to the presence server through the IMS network but  
the

presence server doesn't care about the presence rule documents.

When sending the XCAP PUT request I use the command:

PUT org.openmobilealliance.pres-rules/users/sip:[EMAIL PROTECTED]/pres-rules

no problem so far and the document can also be retreived.

When bob subscribes for alice presence information I can see  
that the

presence server sends a GET request using another AUID:

GET presence-rules/users/sip:[EMAIL PROTECTED]/index

and this results in a 404 error response. Why getting the  
index instead

of pres-rules?

From what I understand the AUID is defined in a different way  
by IETF

and OMA. How can this be handled by OpenIMS/OpenXCAP/OpenSIPS?

Best regards
Kenneth Löfstrand


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Re: [OpenSIPS-Users] Old question about mediaproxy "bridge" mode between public and private networks

[Adrian Georgescu]

Robert,

Could you expand on what you mean by:

1. Privacy
2. Extra security

These seem to be highly abused terms while there is no proper  
description available of what they mean and for whom they provide the  
benefit.


Adrian

On Dec 10, 2008, at 9:32 PM, Robert Dyck wrote:

I see a need for a very basic proxy-like B2BUA. This would  
completely hide the
local topology. This would provide privacy and extra security as  
well as

working around the bad behaviour of some service providers.
Rob

On Wednesday 10 December 2008, Brett Nemeroff wrote:

For what it's worth, I've had problems doing this with some [broken]
carriers. Namely they see a private address in one of the Vias and
they assume it's NAT.. Pretty messy. If you look through the archive
you'll see what happened to me.

That being said, I think it's pretty unusual that this happens.
-Brett

On Wed, Dec 10, 2008 at 8:14 AM, Giuseppe Roberti <[EMAIL PROTECTED]>  
wrote:

Hi.

I have an opensips server running "between" a man local area and
internet. This mean that UAC comes from local area and gateways  
are on

internet.
The local interface (eth0) ip is not reachable from internet.
Opensips server can traverse the nat using add_local_rport(), can
mediaproxy do the same ?

Regards.

--
Giuseppe Roberti
<[EMAIL PROTECTED]>

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Re: [OpenSIPS-Users] Old question about mediaproxy "bridge" mode between public and private networks

[Adrian Georgescu]

Robert,

NAT traversal is solved by OpenSIPS/MediaProxy combination for both  
signalling and media. Cost is important for an operator and any  
intermediate like an SBC, which does not bring any value to end  
customer is not likely to remain there for long.


What I am trying to figure out is if there are other good reasons  
besides the NAT issue for which the insertion of the SBC justifies its  
cost for an operator.


Regards,
Adrian

On Dec 11, 2008, at 2:02 AM, Robert Dyck wrote:

You are right, these terms are used in a rather casual manner. Also  
privacy

and security can never be absolute. However there are reasons why an
individual or organization may want to hide their topology. Those  
with bad

intentions may look for clues so that they may subvert the system.

Perhaps a stronger case can be made when we consider that NAT is  
perhaps the
biggest headache with SIP. Different service providers have  
different ideas
how they might overcome the problem. If a UA on a LAN or an  
extension on a
PBX appears as a simple UA with a public address then the chance of  
success

improves.

OpenSBC may be the way to go. It will act as a proxy or B2BUA. The  
nice thing
about OpenSIPS is its light weight if you don't need a lot of  
modules. I am
not a programmer but it seems to me that it would not be too  
difficult to
hide the private VIAs and CONTACTs. It already supports mediaproxy/ 
rtpproxy.


On Wednesday 10 December 2008, Adrian Georgescu wrote:

Robert,

Could you expand on what you mean by:

1. Privacy
2. Extra security

These seem to be highly abused terms while there is no proper
description available of what they mean and for whom they provide the
benefit.

Adrian

On Dec 10, 2008, at 9:32 PM, Robert Dyck wrote:

I see a need for a very basic proxy-like B2BUA. This would
completely hide the
local topology. This would provide privacy and extra security as
well as
working around the bad behaviour of some service providers.
Rob

On Wednesday 10 December 2008, Brett Nemeroff wrote:
For what it's worth, I've had problems doing this with some  
[broken]

carriers. Namely they see a private address in one of the Vias and
they assume it's NAT.. Pretty messy. If you look through the  
archive

you'll see what happened to me.

That being said, I think it's pretty unusual that this happens.
-Brett

On Wed, Dec 10, 2008 at 8:14 AM, Giuseppe Roberti <[EMAIL PROTECTED]>

wrote:

Hi.

I have an opensips server running "between" a man local area and
internet. This mean that UAC comes from local area and gateways
are on
internet.
The local interface (eth0) ip is not reachable from internet.
Opensips server can traverse the nat using add_local_rport(), can
mediaproxy do the same ?

Regards.

--
Giuseppe Roberti
<[EMAIL PROTECTED]>

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Re: [OpenSIPS-Users] Old question about mediaproxy "bridge" mode between public and private networks

[Adrian Georgescu]

I can only concur with you.

In my experience the trend was always the same. In the beginning the  
operator works hard to implement an SBC for all kind of reasons that  
are not related to his business. After some relative small effort  
everything seem to work in a test environment with a few calls and  
types of phones.


Once real traffic starts flowing, then problems, which were not  
visible in the beginning start to emerge. Once they emerge they only  
expand in scope and multiply in size, which is fine for the SBC  
manufacturer as they have infinite work to process.


The operator then consumes infinite resources navigating around the  
problems introduced by the SBC. In the end everyone wishes to take it  
out but is too late because the architecture is already in place,  
nobody want to admit it was a mistake in the first place, after all it  
was a gigantic vendor selection process where everyone was involved   
and nobody wants to go through the pain of fixing it. Profitability  
has gone done the drain due to over-engineering of the network.


The lesson is to keep the infrastructure simple, make sure you are  
'complying' with whatever regulation is required but don't embed that  
requirement to deep into your product or it will kill it long term.


I wish everyone who starts a SIP business for scratch does not make  
the mistakes many did in the hype VoIP era.


Adrian


On Dec 11, 2008, at 9:36 AM, Brett Nemeroff wrote:


Having a single point of connectivity to the customer, topology
masking,  and potentially CALEA compliance. You can get by without
it.. It's a matter of preference of sorts. Some people have more luck
with nat traversal with them. I'd being interested in hearing other's
experiences with them.

On the other hand, it IS a fixed bottleneck. I can't tell you how many
times I've had a provider's overloaded SBC kill the QOS on my calls..

-Brett


On Thu, Dec 11, 2008 at 2:25 AM, Adrian Georgescu <[EMAIL PROTECTED] 
projects.com> wrote:

Robert,
NAT traversal is solved by OpenSIPS/MediaProxy combination for both
signalling and media. Cost is important for an operator and any  
intermediate
like an SBC, which does not bring any value to end customer is not  
likely to

remain there for long.
What I am trying to figure out is if there are other good reasons  
besides
the NAT issue for which the insertion of the SBC justifies its cost  
for an

operator.
Regards,
Adrian
On Dec 11, 2008, at 2:02 AM, Robert Dyck wrote:

You are right, these terms are used in a rather casual manner. Also  
privacy

and security can never be absolute. However there are reasons why an
individual or organization may want to hide their topology. Those  
with bad

intentions may look for clues so that they may subvert the system.

Perhaps a stronger case can be made when we consider that NAT is  
perhaps the
biggest headache with SIP. Different service providers have  
different ideas
how they might overcome the problem. If a UA on a LAN or an  
extension on a
PBX appears as a simple UA with a public address then the chance of  
success

improves.

OpenSBC may be the way to go. It will act as a proxy or B2BUA. The  
nice

thing
about OpenSIPS is its light weight if you don't need a lot of  
modules. I am
not a programmer but it seems to me that it would not be too  
difficult to
hide the private VIAs and CONTACTs. It already supports mediaproxy/ 
rtpproxy.


On Wednesday 10 December 2008, Adrian Georgescu wrote:

Robert,

Could you expand on what you mean by:

1. Privacy

2. Extra security

These seem to be highly abused terms while there is no proper

description available of what they mean and for whom they provide the

benefit.

Adrian

On Dec 10, 2008, at 9:32 PM, Robert Dyck wrote:

I see a need for a very basic proxy-like B2BUA. This would

completely hide the

local topology. This would provide privacy and extra security as

well as

working around the bad behaviour of some service providers.

Rob

On Wednesday 10 December 2008, Brett Nemeroff wrote:

For what it's worth, I've had problems doing this with some [broken]

carriers. Namely they see a private address in one of the Vias and

they assume it's NAT.. Pretty messy. If you look through the archive

you'll see what happened to me.

That being said, I think it's pretty unusual that this happens.

-Brett

On Wed, Dec 10, 2008 at 8:14 AM, Giuseppe Roberti <[EMAIL PROTECTED]>

wrote:

Hi.

I have an opensips server running "between" a man local area and

internet. This mean that UAC comes from local area and gateways

are on

internet.

The local interface (eth0) ip is not reachable from internet.

Opensips server can traverse the nat using add_local_rport(), can

mediaproxy do the same ?

Regards.

--

Giuseppe Roberti

<[EMAIL PROTECTED]>

___

Re: [OpenSIPS-Users] radius SQL accounting of failed calls

[Adrian Georgescu]

http://download.ag-projects.com/FreeRadius-XS/

Adrian


On Dec 12, 2008, at 9:38 PM, Jeff Pyle wrote:


Hello,

It seems there was a long thread some time back about using the so- 
called non-standard “Acct-Status-Type = Failed” in the radius  
packet.  The two proposed solutions seemed to be 1) use a Stop type  
instead of Failed, or 2) patch Freeradius.


Since I’m still seeing the Failed type in the radius detail file, it  
appears OpenSIPS didn’t change to Stop.


So, anyone know of a patch for current Freeradius sources?


Thanks,
Jeff

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Re: [OpenSIPS-Users] [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

Hello Eric,

To avoid backdoors and possibility of exploiting the Presence server  
code by end-user provisioning we will actually add checks in OpenXCAP  
server for the actual content of RLS services document so that it  
contins only routable sip uris, without any parameters. Also pointers  
to external XCAP documents even if they are mentioned in the RFC will  
no be allowed in our server as it could generate endless loops.


So I encourage you not to use any SIP Uris different than u...@domain  
as you will not be able to manipulate them in the next version.


Regards,
Adrian

On Dec 12, 2008, at 6:13 PM, Eric PTAK wrote:


Hi all,

I'm currently integrating OpenSIPS with OpenXCAP.
After a lot a problem with ubuntu libraries in order to setup  
openxcap, I'm now facing to another issue with RLS module.
I'm using Mercuro and it subsribes to sip:al...@domain;pres-list=Default 
, but the RLS looks for a service at sip:al...@domain so it response  
by a 404 Not Found error.


This is the xml files from xcap :

ep...@rd-srv-devlnx2:~$ xcapclient --app rls-services get
get http://10.26.52.122:8080/xcap-root/rls-services/users/sip:al...@domain/index
etag: "d113935c4c7324c99077a925492251b2"
content-type: application/rls-services+xml
content-length: 467

xmlns="urn:ietf:params:xml:ns:rls-services">

  
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5d


  presence

  


ep...@rd-srv-devlnx2:~$ xcapclient --app resource-lists get
get 
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip:al...@domain/index
etag: "e153e7e4688122a04434b77cd1ecb5e1"
content-type: application/resource-lists+xml
content-length: 324


  
All Contacts
  

  bob


and this is the stack trace :

Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
SUBSCRIBE presence from sip:al...@domain to sip:al...@domain;pres-list=Default 
 (Mercuro IMS Client Beta (4.0.1011.0))
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:presence:search_event: start event= [presence]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:rls_handle_subscribe: 'To' header ALREADY PARSED: >
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:get_resource_list: Searched RL document for user sip:al...@domain
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_new_result: allocate 28 bytes for result set at 0x81b52b0
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: 2 columns returned from the query
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result names at  
0x81b5af8
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result types at  
0x81b5360
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0]  
at 0x81b5820
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81b5820)[0]=[doc]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: use DB_BLOB result type
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[1]  
at 0x81b5838
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81b5838)[1]=[etag]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_convert_rows: allocate 8 bytes for rows at  
0x81b5948
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_convert_row: allocate 40 bytes for row values  
at 0x81b5860
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_str2val: converting BLOB [encoding="utf-8"?>^M ^M   ^M http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5dlist>^M ^M   presence^M packages>^M   ^M ]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_str2val: converting STRING  
[d113935c4c7324c99077a925492251b2]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:get_resource_list: rls_services document: version="1.0" encoding="utf-8"?>^M ^M   ^M http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/

Re: [OpenSIPS-Users] radius SQL accounting of failed calls

[Adrian Georgescu]

On Dec 15, 2008, at 3:25 PM, Jeff Pyle wrote:


Ram,

Be sure to include the list in your reply, not only to me directly.

I’m working towards the same goal you are, on CentOS 5.2 i386.  I  
installed MySQL from RPMs, then unpacked cdrtool_6.6.10.tar.gz into / 
var/www per the instructions.  I unpacked freeradius- 
server-2.1.3.tar.gz into a working area (such as /usr/local/src).  I  
applied the /var/www/CDRTool/contrib/freeradius-brandinger/ 
freeradius_20080103.patch to the freeradius source directory to  
allow the mysterious type-15 “failed” messages to me SQL-accounted.   
I complied freeradius and installed it.


I downloaded unpacked radiusclient-ng-0.5.6.tar.gz, compiled and  
installed it.


At this point for me the trick was to get the conflicting radius  
dictionary attributes settled.  There were some conflicts from the  
freeradius-server includes, the radiusclient-ng dictionary and the  
dictionary.ser included with CDRTool.  I


What are the conflicts, can you specify more details?

Adrian

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Re: [OpenSIPS-Users] [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

  


The above should read sip:al...@domain.com

http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5d


I will not allow this document to be stored in the server as this  
would open Pandera's box of foreign URLs that can be injected in the  
system. The fact that RFC mentioned that is possible does not mean is  
sane to do it as  Presence server operator.


Only  entries containing individual and valid SIP URIs will be  
allowed in rls-services document.


Adrian


On Dec 15, 2008, at 4:53 PM, Eric PTAK wrote:


Thanks you Anca and Adrian for your answer.
Regading the issue with the parameter in the R-URI, I was asking if  
the PS shouldn't use the To header, but I don't find references on   
that any more...


I'll feed back to Mercuro developers in order to remove parameters  
from services URI.


Adrian, when you're talking about external XCAP documents, do you  
mean the use of resource-list tag in rls-services document ?


Eric.



2008/12/15 Adrian Georgescu 
Hello Eric,

To avoid backdoors and possibility of exploiting the Presence server  
code by end-user provisioning we will actually add checks in  
OpenXCAP server for the actual content of RLS services document so  
that it contins only routable sip uris, without any parameters. Also  
pointers to external XCAP documents even if they are mentioned in  
the RFC will no be allowed in our server as it could generate  
endless loops.


So I encourage you not to use any SIP Uris different than  
u...@domain as you will not be able to manipulate them in the next  
version.


Regards,
Adrian

On Dec 12, 2008, at 6:13 PM, Eric PTAK wrote:


Hi all,

I'm currently integrating OpenSIPS with OpenXCAP.
After a lot a problem with ubuntu libraries in order to setup  
openxcap, I'm now facing to another issue with RLS module.
I'm using Mercuro and it subsribes to sip:al...@domain;pres-list=Default 
, but the RLS looks for a service at sip:al...@domain so it  
response by a 404 Not Found error.


This is the xml files from xcap :

ep...@rd-srv-devlnx2:~$ xcapclient --app rls-services get
get http://10.26.52.122:8080/xcap-root/rls-services/users/sip:al...@domain/index
etag: "d113935c4c7324c99077a925492251b2"
content-type: application/rls-services+xml
content-length: 467

xmlns="urn:ietf:params:xml:ns:rls-services">

  
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5d


  presence

  


ep...@rd-srv-devlnx2:~$ xcapclient --app resource-lists get
get 
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip:al...@domain/index
etag: "e153e7e4688122a04434b77cd1ecb5e1"
content-type: application/resource-lists+xml
content-length: 324


  
All Contacts
  

  bob


and this is the stack trace :

Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
SUBSCRIBE presence from sip:al...@domain to sip:al...@domain;pres-list=Default 
 (Mercuro IMS Client Beta (4.0.1011.0))
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:presence:search_event: start event= [presence]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:rls_handle_subscribe: 'To' header ALREADY PARSED: >
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:get_resource_list: Searched RL document for user sip:al...@domain
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_new_result: allocate 28 bytes for result set at 0x81b52b0
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: 2 columns returned from the query
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result names at  
0x81b5af8
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result types at  
0x81b5360
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for  
RES_NAMES[0] at 0x81b5820
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x81b5820)[0]=[doc]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: use DB_BLOB result type
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for  
RES_NAMES[1] at 0x81b5838
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: RES_NA

Re: [OpenSIPS-Users] [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

I would discourage you do both, but this is my opinion. For some good  
practices on using XCAP see this page:


http://openxcap.org/wiki/Running

Adrian

On Dec 15, 2008, at 6:02 PM, Eric PTAK wrote:

And what about if we ensure that the RL is on the same host than the  
rls-services, for example an RLS which is configured like this :


integrated_xcap_server = 0
xcap_root = http://10.26.52.122/xcap-root:8080

In that circumstance, RLS may handle resource-list tag, check if the  
host is equal to xcap_root, and then download the RL.

Do you agree ?

Regards,
Eric.

2008/12/15 Adrian Georgescu 


  


The above should read sip:al...@domain.com

http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5d


I will not allow this document to be stored in the server as this  
would open Pandera's box of foreign URLs that can be injected in the  
system. The fact that RFC mentioned that is possible does not mean  
is sane to do it as  Presence server operator.


Only  entries containing individual and valid SIP URIs will be  
allowed in rls-services document.


Adrian


On Dec 15, 2008, at 4:53 PM, Eric PTAK wrote:


Thanks you Anca and Adrian for your answer.
Regading the issue with the parameter in the R-URI, I was asking if  
the PS shouldn't use the To header, but I don't find references on   
that any more...


I'll feed back to Mercuro developers in order to remove parameters  
from services URI.


Adrian, when you're talking about external XCAP documents, do you  
mean the use of resource-list tag in rls-services document ?


Eric.



2008/12/15 Adrian Georgescu 
Hello Eric,

To avoid backdoors and possibility of exploiting the Presence  
server code by end-user provisioning we will actually add checks in  
OpenXCAP server for the actual content of RLS services document so  
that it contins only routable sip uris, without any parameters.  
Also pointers to external XCAP documents even if they are mentioned  
in the RFC will no be allowed in our server as it could generate  
endless loops.


So I encourage you not to use any SIP Uris different than  
u...@domain as you will not be able to manipulate them in the next  
version.


Regards,
Adrian

On Dec 12, 2008, at 6:13 PM, Eric PTAK wrote:


Hi all,

I'm currently integrating OpenSIPS with OpenXCAP.
After a lot a problem with ubuntu libraries in order to setup  
openxcap, I'm now facing to another issue with RLS module.
I'm using Mercuro and it subsribes to sip:al...@domain;pres-list=Default 
, but the RLS looks for a service at sip:al...@domain so it  
response by a 404 Not Found error.


This is the xml files from xcap :

ep...@rd-srv-devlnx2:~$ xcapclient --app rls-services get
get http://10.26.52.122:8080/xcap-root/rls-services/users/sip:al...@domain/index
etag: "d113935c4c7324c99077a925492251b2"
content-type: application/rls-services+xml
content-length: 467

xmlns="urn:ietf:params:xml:ns:rls-services">

  
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/ 
~~/resource-lists/list...@name=%22default%22%5d


  presence

  


ep...@rd-srv-devlnx2:~$ xcapclient --app resource-lists get
get 
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip:al...@domain/index
etag: "e153e7e4688122a04434b77cd1ecb5e1"
content-type: application/resource-lists+xml
content-length: 324


  
All Contacts
  

  bob


and this is the stack trace :

Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
SUBSCRIBE presence from sip:al...@domain to sip:al...@domain;pres-list=Default 
 (Mercuro IMS Client Beta (4.0.1011.0))
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:presence:search_event: start event= [presence]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:rls_handle_subscribe: 'To' header ALREADY PARSED: >
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:rls:get_resource_list: Searched RL document for user sip:al...@domain
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_new_result: allocate 28 bytes for result set at  
0x81b52b0
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mysql:db_mysql_get_columns: 2 columns returned from the query
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result names at  
0x81b5af8
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:core:db_allocate_columns: allocate 8 bytes for result types at  
0x81b5360
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:  
DBG:db_mys

Re: [OpenSIPS-Users] Opensips-cp: Sip Trace Tool

[Adrian Georgescu]

On Dec 11, 2008, at 8:54 PM, Magnus Burman wrote:

> Hi Matteo,
>
> I ran into the same problem, OpenSips 1.4.3 and Mediaproxy 2.1.0 with
> CDRtool 6.6.10.
>
> In the file /var/www/CDRTool/library/cdr_openser.php I changed
> time_stamp to date on lines 2943 and 3015.

This as well as other issues related to the latest changes in OpenSIPS  
and MediaProxy versions is fixed, a new release will appear soon.

Thanks for the report.

Adrian


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Re: [OpenSIPS-Users] Re : [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

On Dec 15, 2008, at 9:59 PM, Eric PTAK wrote:


Well well, thanks for that great wiki page I didn't see.
What do you think to keep in integrated but use the xcap root module
parameter or other dedicated one to define allowed roots in the
resource-list node ?

Another thing, what about compliance with OMA specifications ?



Which ones more precisely?



Eric

2008/12/15, Adrian Georgescu :

I would discourage you do both, but this is my opinion. For some good
practices on using XCAP see this page:

http://openxcap.org/wiki/Running

Adrian

On Dec 15, 2008, at 6:02 PM, Eric PTAK wrote:


And what about if we ensure that the RL is on the same host than the
rls-services, for example an RLS which is configured like this :

integrated_xcap_server = 0
xcap_root = http://10.26.52.122/xcap-root:8080

In that circumstance, RLS may handle resource-list tag, check if the
host is equal to xcap_root, and then download the RL.
Do you agree ?

Regards,
Eric.

2008/12/15 Adrian Georgescu 


 


The above should read sip:al...@domain.com



http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/

~~/resource-lists/list...@name=%22default%22%5d


I will not allow this document to be stored in the server as this
would open Pandera's box of foreign URLs that can be injected in the
system. The fact that RFC mentioned that is possible does not mean
is sane to do it as  Presence server operator.

Only  entries containing individual and valid SIP URIs will be
allowed in rls-services document.

Adrian


On Dec 15, 2008, at 4:53 PM, Eric PTAK wrote:


Thanks you Anca and Adrian for your answer.
Regading the issue with the parameter in the R-URI, I was asking if
the PS shouldn't use the To header, but I don't find references on
that any more...

I'll feed back to Mercuro developers in order to remove parameters
from services URI.

Adrian, when you're talking about external XCAP documents, do you
mean the use of resource-list tag in rls-services document ?

Eric.



2008/12/15 Adrian Georgescu 
Hello Eric,

To avoid backdoors and possibility of exploiting the Presence
server code by end-user provisioning we will actually add checks in
OpenXCAP server for the actual content of RLS services document so
that it contins only routable sip uris, without any parameters.
Also pointers to external XCAP documents even if they are mentioned
in the RFC will no be allowed in our server as it could generate
endless loops.

So I encourage you not to use any SIP Uris different than
u...@domain as you will not be able to manipulate them in the next
version.

Regards,
Adrian

On Dec 12, 2008, at 6:13 PM, Eric PTAK wrote:


Hi all,

I'm currently integrating OpenSIPS with OpenXCAP.
After a lot a problem with ubuntu libraries in order to setup
openxcap, I'm now facing to another issue with RLS module.
I'm using Mercuro and it subsribes to sip:al...@domain;pres-list=Default

, but the RLS looks for a service at sip:al...@domain so it
response by a 404 Not Found error.

This is the xml files from xcap :

ep...@rd-srv-devlnx2:~$ xcapclient --app rls-services get
get
http://10.26.52.122:8080/xcap-root/rls-services/users/sip:al...@domain/index
etag: "d113935c4c7324c99077a925492251b2"
content-type: application/rls-services+xml
content-length: 467


 

http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/

~~/resource-lists/list...@name=%22default%22%5d
   
 presence
   
 


ep...@rd-srv-devlnx2:~$ xcapclient --app resource-lists get
get
http://10.26.52.122:8080/xcap-root/resource-lists/users/sip:al...@domain/index
etag: "e153e7e4688122a04434b77cd1ecb5e1"
content-type: application/resource-lists+xml
content-length: 324


 
   All Contacts
 
 bob


and this is the stack trace :

Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
SUBSCRIBE presence from sip:al...@domain to
sip:al...@domain;pres-list=Default
(Mercuro IMS Client Beta (4.0.1011.0))
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:core:parse_headers: flags=
Dec 12 15:17:15 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:presence:search_event: start event= [presence]
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:rls:rls_handle_subscribe: 'To' header ALREADY PARSED:



Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:rls:get_resource_list: Searched RL document for user
sip:al...@domain
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:core:db_new_result: allocate 28 bytes for result set at
0x81b52b0
Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:db_mysql:db_mysql_get_columns: 2 columns returned from the  
query

Dec 12 15:17:16 rd-srv-devlnx2 /usr/local/sbin/opensips[25465]:
DBG:core:db_allocate_columns: allocate 8 bytes f

[OpenSIPS-Users] New MediaProxy release 2.3.0

[Adrian Georgescu]

Hello,

There is a new release of MediaProxy software available. To upgrade  
your debian installation:

apt-get update
apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- 
sessions

Or download the tar file from:

http://download.ag-projects.com/MediaProxy/

The changelog since 2.1.0 is below:

mediaproxy (2.3.0) unstable; urgency=low

   * Fixed bug that prevented the dispatcher from trying the next  
relay in
 the list, when a relay was unavailable or unusable
   * Read acctport from the radius config file to an integer not to a  
string
   * Fixed a bug that prevented a call on hold to be resumed in  
certain cases
   * Don't close the media session when on hold and the contrack rule  
expires
   * Enhanced graceful shutdown logic and made it reliable:
 - keep all dispatcher connections active until all sessions are  
gone
 - continue to update the list of active dispatchers
   * Changed function signatures to simplify access to the to_tag  
argument
   * Never return 0.0.0.0 as the IP address to OpenSIPS
   * Fixed a bug with incorrect session counting when errors happen  
while
 creating a new session, which could result in connections with  
obsolete
 dispatchers being kept open indefinitely
   * Made web page display relay summaries even when there are no  
sessions
   * Fixed web page to show correct stream counts when there are no  
sessions
   * Added minimum version build dependendency for libnetfilter- 
conntrack-dev

mediaproxy (2.2.3) unstable; urgency=low

   * Publish software version as part of the network identity
   * Added version command on the dispatcher management interface

mediaproxy (2.2.2) unstable; urgency=low

   * Added minimum pyrad version dependency

mediaproxy (2.2.1) unstable; urgency=low

   * Fixed logic to handle replies from relay to dispatcher on error  
cases

mediaproxy (2.2.0) unstable; urgency=low

   * Clarified debian installation example regarding the testing  
distribution
   * Fixed wording and some typos in the documentation
   * Fixed to_tag not being sent in some cases in synthetic tests
   * Removed Inhibitor object and integrated its functionality into  
the init
 method of ForwardingRule, to improve the session setup efficiency
   * Changed the internal ForwardingRule linked list to a static array
   * Improved the session setup efficiency by 2 orders of magnitude
   * Removed start_time column from database accounting table
   * Renamed pdd to post_dial_delay in media statistics
   * Added management_passport option to the dispatcher
   * Made getting statistics more efficient
   * Added code to get all conntrack rule counters at once for  
efficiency
   * Improved statistics gathering speed more than 2 times
   * Removed redundant top level counters in stats
   * Made speed measurement accurate regarding sessions closed between  
probes
   * Use constant interval scheduler for better accuracy in speed  
measurement
   * Return single line json blobs for summary and sessions
   * Always add dialog_id to session statistics
   * Fixed interaction between dispatcher and opensips for expired  
sessions

Kind regards,
Adrian Georgescu



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[OpenSIPS-Users] Call Control prepaid application for OpenSIPS, new release

[Adrian Georgescu]

Hello,

Call Control is a prepaid application that can be used together with  
OpenSIPS call_control module and CDRTool rating engine to limit the  
duration of SIP sessions based on a prepaid balance. It can also be  
used to limit the duration of any session to a predefined maximum  
value without debiting a balance.

Call Control achieves this by maintaining a timer for each session and  
sending BYE messages to both SIP end-points, if the session exceeds  
its maximum session limit or if the Call Control receives a command to  
forcefully close the call from outside.

Features:

• Parallel sessions using one balance per subscriber
• Support for sessions that have timeout without BYE
• Support for sessions that have timeout for media (using MediaProxy)
• Manual session stop from server terminal
• Overview of ongoing sessions in the web page or server terminal
• Graceful restart without loosing track of sessions
• Detailed logging of all performed actions using syslog
• Web page provisioning for prepaid accounts (using CDRTool)

Download

The software is available as a tar archive at:

http://download.ag-projects.com/CallControl/

For people running Debian testing or unstable there is an official  
public repository. To use it, add these lines in /etc/apt/sources.list

deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

After that, run:

apt-get update apt-get install callcontrol


For more information visit:

http://callcontrol.ag-projects.com


Kind regards,
Adrian Georgescu


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Re: [OpenSIPS-Users] Re : [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

Do you know by any chance what the value '' is support to point to?

pres-list=

Adrian


On Dec 16, 2008, at 10:01 AM, Eric PTAK wrote:

> Hi Adrian,
>
> I'm thinking about OMA-WP-PRS_1_1_Implementation_Guidelines-20080627- 
> C, Implementation Guidelines for OMA Presence SIMPLE v1.1 :
> 5.5.1 Service-URI-Template for Presence Lists
>
> As described in [PRS_RLS_XDM] "
>
> Validation Constraints", the Service URI for Presence Lists (i.e.  
> the value of the "uri"
> attribute of a  element in a Presence List document)  
> proposed by an XDMC when creating a Presence List in the
>
> RLS XDMS must conform to the syntax of the Service-URI-Template  
> parameter described in [PRS_AC] and [PRS_MO].
>
> It is RECOMMENDED that:
>
> ·
>
> the Service-URI-Template for Presence Lists have the following  
> structure:
> ;pres-list=
>
> where the  and  substitution tags are described in  
> [XDM_Core] "
>
> Provisioned XDMC Parameters".
> The reasons for the recommendation include:
>
> ·
>
> effective use of resources (e.g. access network bandwidth), since  
> the recommended Service-URI-Template makes it
> easier for the XDMC to generate a globally unique Service URI that  
> is accepted by the RLS XDMS; and
>
> ·
>
> simplification for the SIP/IP Core network to recognize the Service  
> URI as a Presence List (e.g. to optimize routing
> of Presence List subscriptions).
>
> An example of a Service URI conforming to the recommended Service- 
> URI-Template is as follows:
>
> sip:j...@example.com;pres-list=list-a
>
> where the XUI used to generate the Service URI is a SIP URI, as  
> required by [PRS_RLS_XDM] "
>
> Validation constraints".
>
>
> Regards,
> Eric.
>
> 2008/12/15 Adrian Georgescu 
>
> On Dec 15, 2008, at 9:59 PM, Eric PTAK wrote:
>
>> Well well, thanks for that great wiki page I didn't see.
>> What do you think to keep in integrated but use the xcap root module
>> parameter or other dedicated one to define allowed roots in the
>> resource-list node ?
>>
>> Another thing, what about compliance with OMA specifications ?
>>
>
> Which ones more precisely?
>
>
>> Eric
>>
>> 2008/12/15, Adrian Georgescu :
>>> I would discourage you do both, but this is my opinion. For some  
>>> good
>>> practices on using XCAP see this page:
>>>
>>> http://openxcap.org/wiki/Running
>>>
>>> Adrian
>>>
>>> On Dec 15, 2008, at 6:02 PM, Eric PTAK wrote:
>>>
>>>> And what about if we ensure that the RL is on the same host than  
>>>> the
>>>> rls-services, for example an RLS which is configured like this :
>>>>
>>>> integrated_xcap_server = 0
>>>> xcap_root = http://10.26.52.122/xcap-root:8080
>>>>
>>>> In that circumstance, RLS may handle resource-list tag, check if  
>>>> the
>>>> host is equal to xcap_root, and then download the RL.
>>>> Do you agree ?
>>>>
>>>> Regards,
>>>> Eric.
>>>>
>>>> 2008/12/15 Adrian Georgescu 
>>>>
>>>>>  
>>>>
>>>> The above should read sip:al...@domain.com
>>>>
>>>>>
>>>>> http://10.26.52.122:8080/xcap-root/resource-lists/users/sip%3aalice%40domain/index/
>>>>>
>>>>> ~~/resource-lists/list...@name=%22default%22%5d
>>>>
>>>> I will not allow this document to be stored in the server as this
>>>> would open Pandera's box of foreign URLs that can be injected in  
>>>> the
>>>> system. The fact that RFC mentioned that is possible does not mean
>>>> is sane to do it as  Presence server operator.
>>>>
>>>> Only  entries containing individual and valid SIP URIs will be
>>>> allowed in rls-services document.
>>>>
>>>> Adrian
>>>>
>>>>
>>>> On Dec 15, 2008, at 4:53 PM, Eric PTAK wrote:
>>>>
>>>>> Thanks you Anca and Adrian for your answer.
>>>>> Regading the issue with the parameter in the R-URI, I was asking  
>>>>> if
>>>>> the PS shouldn't use the To header, but I don't find references on
>>>>> that any more...
>>>>>
>>>>> I'll feed back to Mercuro developers in order to remove parameters
>>>>> from services URI.
>>>>>
>>>>> Adrian, when you're tal

Re: [OpenSIPS-Users] Freeradius' schema.sql breaks CDRTool

[Adrian Georgescu]

Yes, use that file CDRTool/setup/radius/OpenSIPs/radacct.mysql

Adrian


On Dec 18, 2008, at 8:58 PM, Jeff Pyle wrote:


Hello,

CDRTool’s INSTALL.txt says to use /usr/share/doc/freeradius/examples/ 
db_mysql.sql.gz to create the Radius backend framework for DB  
storage.  The closest file available in freeradius-server-2.1.3 is / 
usr/local/etc/raddb/sql/mysql/schema.sql.  Unfortunately it creates  
all the field names in lowercase.  While this doesn’t affect Radius,  
it does prevent CDRTool from querying much of the data.  Perhaps  
using CDRTool/setup/radius/OpenSIPs/radacct.mysql might be a better  
option?  It looks like radacct.mysql contains all the updates from  
radacct-patch.sh as well.


- Jeff
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[OpenSIPS-Users] new CDRTool release 6.7.1

[Adrian Georgescu]

Hello,

The software contains improvements for compatibility with the latest  
OpenSIPS and MediaProxy versions and the installation documentation.  
An installation web page has been created at:

http://cdrtool.ag-projects.com/wiki/Install

Changelog

cdrtool (6.7.1) unstable; urgency=low

   * Simplified the sample configuration file global.inc.simple.sample
   * Improved installation documentation

cdrtool (6.7.0) unstable; urgency=low

   * Migrated to OpenSIPS, this version is incompatible with OpenSER,  
do not
 upgrade to this version if you still use OpenSER
   * Added support for Call Control prepaid application from
 http://download.ag-projects.com/CallControl
   * Normalize the duration of prepaid CDRs to match the duration  
calculated
 by Call Control, this avoids price missmatch between debit price  
and
 the price calculated during normalization
   * Updated OpenSIPS radius dictionary
   * Updated Freeradius server setup procedure with OpenSIPS and  
MediaProry
 dictionaries
   * Added checks for values in rating tables changed from the web page
   * Improved retrieval and display of SIP traces from OpenSIPS
   * Improved retrieval and display of Media traces from MediaProxy
   * Reset the imported records counter before importing each csv file
   * Added new fields in billing_rates connectCostIn and  
durationRateIn to
 store the purchased price, the rating info now contains two  
prices for
 each call. The CSV format of rating table has been changed to  
support
 provisioning of purchased prices. Updated sample csv file for
 billing_rates.
   * Improved rating documentationin general
   * Moved E164 class to cdr_generic.php and allow specification of  
which
 E164 class should be used for each datasource
   * Updated sample configuration files in setup/global.inc.*.sample
   * Added new columns to prepaid_history to store information about  
sessions
 that lead to debit balance
   * Log duration of prepaid sessions in prepaid_history table
   * Removed unused trafficRate from the rating engine and mysql tables
   * Added CallerId to SIP accounts in ENUM generator
   * Added support for new LCR engine from NGNPro4
   * Removed one empty column from ENUM records that had no mappings
   * Added domain column to the prepaid table
   * To upgrade you must apply the changes from doc/Upgrade.txt and
 setup/mysql/alter_tables.mysql


The software can be downloaded from:

http://download.ag-projects.com/CDRTool/

For people running Debian testing there is an official public  
repository. To use it, add
these lines in /etc/apt/sources.list

# AG Projects software
deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

Install the AG Projects debian software signing key:

wget http://download.ag-projects.com/agp-debian-gpg.key
apt-key add agp-debian-gpg.key

After that, run:

apt-get update
apt-get install cdrtool

Regards,
Adrian


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Re: [OpenSIPS-Users] Upgrade CDRTool to 6.7.1 - problems

[Adrian Georgescu]

Carlo,

From your description it looks like an incorrect configuration in  
global.inc.


If I were you I would start with a fresh  global.inc.simple.sample and  
port the settings from the old version.


Adrian


On Dec 29, 2008, at 3:41 AM, Carlo Dimaggio wrote:


Anybody can help me?
Where I can find more informations about this "datasource ERROR"?

Thank you!


Il giorno 23/dic/08, alle ore 22:23, Carlo Dimaggio ha scritto:


Hi all,

I have some problems with the upgrade of CDRTool from 6.6.10 to  
6.7.1.

I have downloaded the debian package (my machine is an ubuntu server)
and followed the instruction included in 
http://download.ag-projects.com/CDRTool/doc/Upgrade.txt
.
Now I can start (and reload) successfully cdrtool but if I connect to
the web interface this error is shown:

"Error initializing CDRTool datasource sip_trace"

Before the upgrade all was fine (mysql user/pass, permissions,
configurations,...). There are no errors in syslog.

How can I debug the problem?

Thanks,
Carlo


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Re: [OpenSIPS-Users] Re : [RLS] Cannot found service uri in rls-services

[Adrian Georgescu]

Hi Eric,

I added a few notes about this issue here:

http://openxcap.org/wiki/Running

Section "RLS services checks"

Comments are welcome to capture the relevant requirements.

Regards,
Adrian


On Dec 30, 2008, at 5:49 AM, Eric PTAK wrote:

> Hi everyones,
>
> I hope you spend a merry christmas and you're ready for the new  
> years eve ;)
>
> As I'm still working on RLS subscriptions, I'm modding the module to  
> allow resource-lists references within rls-services doc.
> I will made something with an option to allow only certains xcap  
> roots, and I'll first check DB (and only it if integrated xcap).
> I would like to know more about a future support of pres-list  
> parameter of SUBSCRIBEs R-URI.
> Did you discuss about that ? Has it been roadmapped ?
>
> Have a nice saint-sylvestre,
> Eric.
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Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

[Adrian Georgescu]

I do not find in your logs what is wrong with your setup, it could be  
either the client or the server configuration.


Can you try your XCAP client using a SIP account from http://sip2sip.info 
 see if that works?


Regards,
Adrian


On Jan 2, 2009, at 2:39 PM, Jun.Wen wrote:


It looks like this -

-
xcap:~/.sipclient# cat config.ini
# rename this file to 'config.ini' and copy it in ~/.sipclient/  
directory


# this will be the default account used by xcapclient
[Account]
sip_address = al...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

# this will be used when -a bob command-line switch is provided
[Account_bob]
sip_address = b...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月2日 21:35
To: Jun.Wen
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What is the content of your xcapclient configuration file .sipclient/ 
config.ini ?


Adrian

On Jan 2, 2009, at 9:33 AM, Jun.Wen wrote:


Hi, All, Happy new year to all the team here.

Please help me to figure out what is going wrong with my openxcap.  
Thanks in advance.


I've built up openxcap in debian according to the installation  
guide and the it works as followings info from error.log.

---
2009-01-02 02:56:26-0500 [-] Starting OpenXCAP 1.0.6
2009-01-02 02:56:37-0500 [-] Supported Root URIs: http://192.168.10.10/xcap-root
2009-01-02 02:56:39-0500 [-] Certificate file 'tls/server.crt'  
could not be loaded: File 'tls/server.crt' does not exist
2009-01-02 02:56:39-0500 [-] Private key file 'tls/server.key'  
could not be loaded: File 'tls/server.key' does not exist
2009-01-02 02:56:39-0500 [-] Trusted peers: 192.168.10.0/24,  
127.0.0.1

2009-01-02 02:56:39-0500 [-] xcap.server.HTTPFactory starting on 80


The openxcap server is configured to domain 192.168.10.10 ( local  
server static IP address ). I also created a test account al...@192.168.10.10 
 with pwd 123 by python add-openxcap-user.py in scripts.  While  
when I tried the python-xcapclient to put a document to the  
openxcap server, I always encountered "401 Unauthorized" for  
following details -

---
xcap:/opt/python-xcaplib/examples# xcapclient -i pres-rules.xml put
put 
http://192.168.10.10/xcap-root/pres-rules/users/sip:al...@192.168.10.10/index
401 Unauthorized
content-type: text/html
content-length: 141
Unauthorizedhead>UnauthorizedYou are not authorized to access  
this resource.

---

Please refer the the access.log and it seems something related my  
domain settings.


---
2009-01-01 08:41:08-0500 [-] 192.168.10.10 'PUT /xcap-root/pres- 
rules/users/sip:al...@192.168.10.10/index HTTP/1.1' 401 0 141  
'python-xcaplib/1.0.8' -

REQUEST headers:
Accept-Encoding: identity
User-Agent: python-xcaplib/1.0.8
Host: 192.168.10.10
Content-Type: application/x-www-form-urlencoded
Authorization: Basic YWxpY2U6MTIz
RESPONSE headers:
Date: Thu, 01 Jan 2009 13:41:08 GMT
Content-Type: text/html
WWW-Authenticate: basic realm="192.168.10.10"
Server: OpenXCAP/1.0.6
---

Regards

Jun






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Re: [OpenSIPS-Users] opensips on centos 64

[Adrian Georgescu]

You can use the RPC proxy instead of the bultin xmlrpc server, which  
is not reliable as is freezing out of the blue. You need to install  
Python for it.


Your can find the proxy it here:

openser-mi-proxy-1.0.0.tar.gz

Adrian


On Jan 5, 2009, at 6:21 PM, J Santos wrote:

There is no such package available in my system. That's why I am  
trying to

build it from rpm source without success.

J. Santos








-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
Sent: Monday, January 05, 2009 3:16 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64


Hi,

Try using the xmlrpc-c3 package available on your system (whatever
version it is) and ignore the compile warning from the
mi_xmlrpc module.

Regards,
Bogdan

J Santos wrote:

I am retrying this so may be someone out there could kindly take a
look and provide me some directions.

I am using Centos 64 and I am having difficulties to make

xmlrpc-c to

work. Version 0.9.1.10  doesn't work on 64 bits so I am trying to
create a rpm from rpm-source.

When rpmbuild -ba xmlrpc.spec  I am getting 'machine x86_64 is not
recognized'.

Does anybody out there succesfully installed opensips in a 64 bit
machine ?

thanks

Jair Santos








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Checked by AVG - http://www.avg.com
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Date: 1/3/2009 2:14 PM




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Re: [OpenSIPS-Users] setup openxcap

[Adrian Georgescu]

You must install the dependencies on your system. They are described  
here:


http://openxcap.org/wiki/Installation

Adrian


On Dec 24, 2008, at 7:05 PM, troxlinux wrote:



somebody in the list with a help..

Stopping OpenXCAP server: openxcap .
Starting OpenXCAP server: openxcap Traceback (most recent call last):
  File "/usr/local/bin//openxcap", line 41, in 
from xcap.logutil import start_log
  File "/usr/local/lib/python2.5/site-packages/xcap/logutil.py",  
line 4, in 

from twisted.web2 import responsecode
ImportError: No module named web2


--
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Re: [OpenSIPS-Users] opensips on centos 64

[Adrian Georgescu]

I updated the Installation guide here:

http://openxcap.org/wiki/Installation

Regards,
Adrian


On Jan 5, 2009, at 7:11 PM, J Santos wrote:


Thank you Adrian,


I downloaded the package but I don't know how to install / load it.

I couldn't find enough information in the package or in the archive.

Is there any documentation available ?

-- JS

-Original Message-----
From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: Monday, January 05, 2009 9:30 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64

You can use the RPC proxy instead of the bultin xmlrpc server, which  
is not reliable as is freezing out of the blue. You need to install  
Python for it.


Your can find the proxy it here:

openser-mi-proxy-1.0.0.tar.gz

Adrian


On Jan 5, 2009, at 6:21 PM, J Santos wrote:

There is no such package available in my system. That's why I am  
trying to

build it from rpm source without success.

J. Santos








-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
Sent: Monday, January 05, 2009 3:16 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64


Hi,

Try using the xmlrpc-c3 package available on your system (whatever
version it is) and ignore the compile warning from the
mi_xmlrpc module.

Regards,
Bogdan

J Santos wrote:

I am retrying this so may be someone out there could kindly take a
look and provide me some directions.

I am using Centos 64 and I am having difficulties to make

xmlrpc-c to

work. Version 0.9.1.10  doesn't work on 64 bits so I am trying to
create a rpm from rpm-source.

When rpmbuild -ba xmlrpc.spec  I am getting 'machine x86_64 is not
recognized'.

Does anybody out there succesfully installed opensips in a 64 bit
machine ?

thanks

Jair Santos








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[OpenSIPS-Users] SIP URI for presence testing: al...@ag-projects.com

[Adrian Georgescu]

Hello,

For development and testing of presence clients and servers I setup a  
test SIP address. You can test by sending a SUBSCRIBE for event  
presence to sip:al...@ag-projects.com, you will receive NOTIFY  
messages back.


Alice's account has a random state generator that publishes state  
changes every minute. The PIDF contains Person, Device and Service  
with information like place-is, mood, notes and activities:


Subscribing to "sip:al...@ag-projects.com" for the presence event, at 
udp:85.17.186.7:5060


Received NOTIFY:

Presence for al...@ag-projects.com:
  Person id: tquafqbd
Timestamp: 2009-01-03 15:25:09
Note(en): You may worry about your hair-do today, but tomorrow  
much peanut butter will be sold.

Mood: neutral
Place information:
  Audio: ok
  Video: dark
  Text: uncomfortable
Time offset from UTC: 60 minutes
  ---
  Service id: curvcjxx
Timestamp: 2009-01-03 15:25:09
Status: closed
Contact: sip:al...@ag-projects.com
Relationship: self
  ---
  Device id: gcdzvwje
Timestamp: 2009-01-03 15:10:09
Note(en): Powered by ag-projects/sipclient-0.3.0-pjsip-1.0.1


I hope this is found useful.

Regards,
Adrian

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Re: [OpenSIPS-Users] opensips on centos 64

[Adrian Georgescu]

For openser-mi-proxy the dependencies are:

python-application (>= 1.0.9)
python-twisted-core
python-twisted-web2

Python 2.4 is OK

Regards,
Adrian


On Jan 5, 2009, at 10:00 PM, J Santos wrote:


Hello again Adrian,

one of the requirements is Python 2.5 or newer. It is very clear but  
it doesn't hurt to ask . Is there any chance that I could keep my  
2.4.3 version ?


If not , there is the following note on version 2.5 installer that  
could affect me. Do I need any of these modules ?



"64-bit platforms: The modules audioop, imageop and rgbimg don't work.
The setup.py script disables them on 64-bit installations.
Don't try to enable them in the Modules/Setup file.  They
contain code that is quite wordsize sensitive.  (If you have a
fix, let us know!)"
thanks

JS



-Original Message-
From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: Monday, January 05, 2009 11:39 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64

I updated the Installation guide here:

http://openxcap.org/wiki/Installation

Regards,
Adrian


On Jan 5, 2009, at 7:11 PM, J Santos wrote:


Thank you Adrian,


I downloaded the package but I don't know how to install / load it.

I couldn't find enough information in the package or in the archive.

Is there any documentation available ?

-- JS


-----Original Message-
From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: Monday, January 05, 2009 9:30 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64

You can use the RPC proxy instead of the bultin xmlrpc server,  
which is not reliable as is freezing out of the blue. You need to  
install Python for it.


Your can find the proxy it here:

openser-mi-proxy-1.0.0.tar.gz

Adrian


On Jan 5, 2009, at 6:21 PM, J Santos wrote:

There is no such package available in my system. That's why I am  
trying to

build it from rpm source without success.

J. Santos








-Original Message-
From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
Sent: Monday, January 05, 2009 3:16 AM
To: J Santos
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] opensips on centos 64


Hi,

Try using the xmlrpc-c3 package available on your system (whatever
version it is) and ignore the compile warning from the
mi_xmlrpc module.

Regards,
Bogdan

J Santos wrote:

I am retrying this so may be someone out there could kindly take a
look and provide me some directions.

I am using Centos 64 and I am having difficulties to make

xmlrpc-c to

work. Version 0.9.1.10  doesn't work on 64 bits so I am trying to
create a rpm from rpm-source.

When rpmbuild -ba xmlrpc.spec  I am getting 'machine x86_64 is not
recognized'.

Does anybody out there succesfully installed opensips in a 64 bit
machine ?

thanks

Jair Santos








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Checked by AVG - http://www.avg.com
Version: 8.0.176 / Virus Database: 270.10.2/1873 - Release
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Checked by AVG - http://www.avg.com
Version: 8.0.176 / Virus Database: 270.10.2/1873 - Release Date:  
1/3/2009 2:14 PM




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Checked by AVG - http://www.avg.com
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Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

[Adrian Georgescu]

Did you try opensipsctl command?

Adrian

On Jan 7, 2009, at 5:33 AM, Jun.Wen wrote:


Adrian,

I guess my problem is I did not successfully created users in xcap  
server by the scripts of "python add-openxcap-user.py". Inside the  
table of subscriber of mysql, there is no any user record existed.


Any tips to create user inside xcap server ?

Regards

Jun

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月3日 1:08
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

I do not find in your logs what is wrong with your setup, it could  
be either the client or the server configuration.


Can you try your XCAP client using a SIP account from http://sip2sip.info 
 see if that works?


Regards,
Adrian


On Jan 2, 2009, at 2:39 PM, Jun.Wen wrote:


It looks like this -

-
xcap:~/.sipclient# cat config.ini
# rename this file to 'config.ini' and copy it in ~/.sipclient/  
directory


# this will be the default account used by xcapclient
[Account]
sip_address = al...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

# this will be used when -a bob command-line switch is provided
[Account_bob]
sip_address = b...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月2日 21:35
To: Jun.Wen
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What is the content of your xcapclient configuration  
file .sipclient/config.ini ?


Adrian

On Jan 2, 2009, at 9:33 AM, Jun.Wen wrote:


Hi, All, Happy new year to all the team here.

Please help me to figure out what is going wrong with my openxcap.  
Thanks in advance.


I've built up openxcap in debian according to the installation  
guide and the it works as followings info from error.log.

---
2009-01-02 02:56:26-0500 [-] Starting OpenXCAP 1.0.6
2009-01-02 02:56:37-0500 [-] Supported Root URIs: http://192.168.10.10/xcap-root
2009-01-02 02:56:39-0500 [-] Certificate file 'tls/server.crt'  
could not be loaded: File 'tls/server.crt' does not exist
2009-01-02 02:56:39-0500 [-] Private key file 'tls/server.key'  
could not be loaded: File 'tls/server.key' does not exist
2009-01-02 02:56:39-0500 [-] Trusted peers: 192.168.10.0/24,  
127.0.0.1

2009-01-02 02:56:39-0500 [-] xcap.server.HTTPFactory starting on 80


The openxcap server is configured to domain 192.168.10.10 ( local  
server static IP address ). I also created a test account al...@192.168.10.10 
 with pwd 123 by python add-openxcap-user.py in scripts.  While  
when I tried the python-xcapclient to put a document to the  
openxcap server, I always encountered "401 Unauthorized" for  
following details -

---
xcap:/opt/python-xcaplib/examples# xcapclient -i pres-rules.xml put
put 
http://192.168.10.10/xcap-root/pres-rules/users/sip:al...@192.168.10.10/index
401 Unauthorized
content-type: text/html
content-length: 141
Unauthorizedhead>UnauthorizedYou are not authorized to  
access this resource.

---

Please refer the the access.log and it seems something related my  
domain settings.


---
2009-01-01 08:41:08-0500 [-] 192.168.10.10 'PUT /xcap-root/pres- 
rules/users/sip:al...@192.168.10.10/index HTTP/1.1' 401 0 141  
'python-xcaplib/1.0.8' -

REQUEST headers:
Accept-Encoding: identity
User-Agent: python-xcaplib/1.0.8
Host: 192.168.10.10
Content-Type: application/x-www-form-urlencoded
Authorization: Basic YWxpY2U6MTIz
RESPONSE headers:
Date: Thu, 01 Jan 2009 13:41:08 GMT
Content-Type: text/html
WWW-Authenticate: basic realm="192.168.10.10"
Server: OpenXCAP/1.0.6
---

Regards

Jun






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Re: [OpenSIPS-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?

[Adrian Georgescu]

I beg to differ, but this is just my humble opinion based on my  
experience with my particular customers.


The most economic and future-proof way to perform accounting for SIP  
sessions is the SIP Proxy server alone.


My personal experience is that gateways come and go in a provider  
configuration and they are in many cases under the control of a third- 
party that provides the PSTN termination service. When you do LCR  
across many different gateways, which are not even yours the only  
aggregation point for traffic is the SIP proxy that authenticates and  
authorizes the requests. Over time, the gateways change hands, get  
upgraded or removed much more often then the proxy itself, which  
maintains its central role over time. Secondly, once you do more the  
voice like video and other services that require billing and are not  
PSTN related, the SIP Proxy is the only network element that has  
access to the signalling and is able to generate accounting tickets.


Solving the accounting related problems at the SIP Proxy level is a  
worthwhile investment while other options are just temporary fixes  
that work in a particular case for a limited amount of time and that  
is a waste of money.


Adrian

On Jan 7, 2009, at 2:25 AM, Jiri Kuthan wrote:


authentication does not provide actually value here. dialog would not
either, since
the same trick can be achieved for example by low max-forwards. IMO  
the

proper
choice is accounting from the gateway, which provides the actual  
service.

A proxy can only provide an approximation which is inherentely to some
extent
more error-prone than the box doing the actual job.

-jiri

Bogdan-Andrei Iancu wrote:

Hi Iñaki,

Have you consider requesting auth for the BYE ? from SIP point of  
view

is perfectly valid

Regards,
Bogdan

Iñaki Baz Castillo wrote:

Hi, I'm thinking in the following flow in which the caller/attacker
would get an unlimited call (but a limited CDR duration):

--
attacker OpenSIPS (Acc) 
gateway


INVITE (CSeq 12)  -->
< 407 Proxy Auth

INVITE (CSeq 13)  -->
 INVITE (CSeq 13)   
-->
 <---  
200 Ok

<--- 200 Ok
 << Acc START >>
ACK (CSeq 13) --->
 ACK (CSeq 13)  
--->


<*** RTP >

# Fraudulent BYE !!!
BYE (CSeq 10) --->
 << Acc STOP >>
 BYE (CSeq 10)  
--->
 <-- 500 Req Out of  
Order

<-- 500 Req Out of Order
--

The call hasn't finished, but OpenSIPS has ended the accounting for
this call since it received a BYE. And this BYE will generate a
correct ACC Stop action (since it matches From_tag, To_tag and
Call-ID).

I think this is *VERY* dangerous and I hope I'm wrong.

Would help the dialog module here? does the dialog module check the
CSeq of the BYE in some way and could it prevent OpenSIPS from
generating the ACC STOP action? (I don't think so).

Any idea?








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Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

[Adrian Georgescu]

The database is commonly shared between the two servers. So you can  
create an user with opensipsctl command on the sip machine as long as  
OpenXCAP is using the same database as OpenSIPS.


Adrian

On Jan 7, 2009, at 8:29 AM, Jun.Wen wrote:

I did not install opensips with openxcap in same server, is it a  
must or not ?


From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月7日 14:49
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

Did you try opensipsctl command?

Adrian

On Jan 7, 2009, at 5:33 AM, Jun.Wen wrote:


Adrian,

I guess my problem is I did not successfully created users in xcap  
server by the scripts of "python add-openxcap-user.py". Inside the  
table of subscriber of mysql, there is no any user record existed.


Any tips to create user inside xcap server ?

Regards

Jun

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月3日 1:08
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

I do not find in your logs what is wrong with your setup, it could  
be either the client or the server configuration.


Can you try your XCAP client using a SIP account from http://sip2sip.info 
 see if that works?


Regards,
Adrian


On Jan 2, 2009, at 2:39 PM, Jun.Wen wrote:


It looks like this -

-
xcap:~/.sipclient# cat config.ini
# rename this file to 'config.ini' and copy it in ~/.sipclient/  
directory


# this will be the default account used by xcapclient
[Account]
sip_address = al...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

# this will be used when -a bob command-line switch is provided
[Account_bob]
sip_address = b...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月2日 21:35
To: Jun.Wen
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What is the content of your xcapclient configuration  
file .sipclient/config.ini ?


Adrian

On Jan 2, 2009, at 9:33 AM, Jun.Wen wrote:


Hi, All, Happy new year to all the team here.

Please help me to figure out what is going wrong with my  
openxcap. Thanks in advance.


I've built up openxcap in debian according to the installation  
guide and the it works as followings info from error.log.

---
2009-01-02 02:56:26-0500 [-] Starting OpenXCAP 1.0.6
2009-01-02 02:56:37-0500 [-] Supported Root URIs: http://192.168.10.10/xcap-root
2009-01-02 02:56:39-0500 [-] Certificate file 'tls/server.crt'  
could not be loaded: File 'tls/server.crt' does not exist
2009-01-02 02:56:39-0500 [-] Private key file 'tls/server.key'  
could not be loaded: File 'tls/server.key' does not exist
2009-01-02 02:56:39-0500 [-] Trusted peers: 192.168.10.0/24,  
127.0.0.1

2009-01-02 02:56:39-0500 [-] xcap.server.HTTPFactory starting on 80


The openxcap server is configured to domain 192.168.10.10 ( local  
server static IP address ). I also created a test account al...@192.168.10.10 
 with pwd 123 by python add-openxcap-user.py in scripts.  While  
when I tried the python-xcapclient to put a document to the  
openxcap server, I always encountered "401 Unauthorized" for  
following details -

---
xcap:/opt/python-xcaplib/examples# xcapclient -i pres-rules.xml put
put 
http://192.168.10.10/xcap-root/pres-rules/users/sip:al...@192.168.10.10/index
401 Unauthorized
content-type: text/html
content-length: 141
Unauthorizedhead>UnauthorizedYou are not authorized to  
access this resource.

---

Please refer the the access.log and it seems something related my  
domain settings.


---
2009-01-01 08:41:08-0500 [-] 192.168.10.10 'PUT /xcap-root/pres- 
rules/users/sip:al...@192.168.10.10/index HTTP/1.1' 401 0 141  
'python-xcaplib/1.0.8' -

REQUEST headers:
Accept-Encoding: identity
User-Agent: python-xcaplib/1.0.8
Host: 192.168.10.10
Content-Type: application/x-www-form-urlencoded
Authorization: Basic YWxpY2U6MTIz
RESPONSE headers:
Date: Thu, 01 Jan 2009 13:41:08 GMT
Content-Type: text/html
  

Re: [OpenSIPS-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?

[Adrian Georgescu]

On Jan 7, 2009, at 9:47 AM, Jiri Kuthan wrote:


Adrian Georgescu wrote:
I beg to differ, but this is just my humble opinion based on my  
experience with my particular customers.
The most economic and future-proof way to perform accounting for  
SIP sessions is the SIP Proxy server alone.


This may be probably ok, as long as you don't intend to use such  
accounting

data for billing. (which may be still useful)



I really mean accounting for billing purpose.

The trouble is that proxy-produced accounting data is remarkable  
incomplete and
inaccurate. It does not include QoS info, PSTN info, and they are  
sensitive to
the attacks mentioned  before that make a BYE work for a GW but not  
for a proxy
and vice versa, or other ways how BYE can be broken due to an error  
or fraud.


Again these are issue that need to be addressed and do not imply that  
SIP Proxy accounting is not possible or undesirable.


My personal experience is that gateways come and go in a provider  
configuration and they are in many cases under the control of a  
third-party that provides the PSTN termination service. When you do  
LCR across many different gateways, which are not even yours the  
only aggregation point for traffic is the SIP proxy that  
authenticates and authorizes the requests. Over time, the gateways  
change hands, get upgraded or removed much more often then the  
proxy itself, which maintains its central role over time.


There is certainly some invariable in a system but to my best  
knowledge
that's the DB backend (for example RADIUS) which gets almost never  
touched,
not a proxy server. The DB is the piece that is invariable,  
untouchable,
central in every respect, and therefore used for aggregation of  
usage data,
as directly as possible. I see little value on putting a SIP proxy  
on the
way from the service box knowing ALL call data and the final  
destination

of the usage data (some database).


When I referred to the accounting of the SIP Proxy server my intention  
was to denominate "The accounting server (like Radius) associated with  
the SIP Proxy and its DB backend" as in your example. So we talk about  
the same thing.


(I agree proxy is the best place for authorization and  
authentication but that's

a different story than accounting.)


Secondly, once you
do more the voice like video and other services that require  
billing and are not PSTN related, the SIP Proxy is the only network  
element that has access to the signalling and is able to generate  
accounting tickets.


That seems appealing indeed, it is just that I have encountered very  
few (still some
though) who would be seriously billing for on-net calls on a per- 
minute basis.

(they haven't found a way to do sell credibly a single usrloc lookup
on a per-minute basis or didn't consider the on-net share of traffic  
significant or
thought the CDR producing expense was just not worth it) It makes  
sense as you say
to produce CDRs in a proxy if termination is provided by a third  
party, but to my
best knowledge these are based on their inaccuracy used for  
reconciliation rather

than as source of authoritative data.


There are SIP service numbers that are not available on PSTN and  
charged per minute like PSTN destinations. Then there are peering  
agreements that allow calls to be routed based on results of ENUM  
lookups and still charged per minute. No gateway involved just an ENUM  
lookup. Only the SIP Proxy knows this information.


Solving the accounting related problems at the SIP Proxy level is a  
worthwhile investment



Yes, but only if you don't care about accuracy and completeness of  
the usage data,
i.e., you don't do billing. Otherwise the customer-care cost is  
unpayable in addition
to the expense of doing it at all. The per-minute margins are so  
poor and accurate

CDR processing is such an expense,


I beg to differ. The accounting can be as accurate as any other source  
like the PSTN gateway if you consider that a relevant comparison  
factor. The fact that a particular implementation does not address the  
flaws mentioned here does not rule out SIP Proxy accounting is not good.



that it alone explains the increasing flat-rate
offerings. We have been doing it only in the reconciliation case you  
mentioned,

merely as non-authoritative data.

If you however do have a scenario, in which accuracy and  
completeness matters for
billing sake, investment in proxy-based CDR production seems to me  
only likely to

produce liability.



So is no debate here.



-jiri


while other options are just temporary fixes that
work in a particular case for a limited amount of time and that is  
a waste of money.

Adrian
On Jan 7, 2009, at 2:25 AM, Jiri Kuthan wrote:
authentication does not provide actually value here. dialog would  
not

either, since
the same trick can be achieved for example by low max-forwards.  
IMO the

proper
choice is accounting from the ga

Re: [OpenSIPS-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?

[Adrian Georgescu]

The dialog module could eventually be used to detect out of sync Cseq  
and take decision to terminate the call. Is this feasible?


Adrian

On Dec 19, 2008, at 3:59 PM, Victor Pascual Ávila wrote:


On Fri, Dec 19, 2008 at 3:22 PM, Bogdan-Andrei Iancu
 wrote:

Hi Iñaki,

Have you consider requesting auth for the BYE ? from SIP point of  
view

is perfectly valid


I'm afraid this would only prevent external attackers but does not
protect you from your own customers-- guys who have the credentials
and wanna call for free.

Cheers,
--
Victor Pascual Ávila
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Re: [OpenSIPS-Users] Accounting: How to avoid a fraudulent BYE with lower CSeq?

[Adrian Georgescu]

On Jan 7, 2009, at 2:28 PM, Klaus Darilion wrote:




Adrian Georgescu schrieb:
I beg to differ, but this is just my humble opinion based on my  
experience with my particular customers.
The most economic and future-proof way to perform accounting for  
SIP sessions is the SIP Proxy server alone.
My personal experience is that gateways come and go in a provider  
configuration and they are in many cases under the control of a  
third-party that provides the PSTN termination service. When you do  
LCR across many different gateways, which are not even yours the  
only aggregation point for traffic is the SIP proxy that  
authenticates and authorizes the requests. Over time, the gateways  
change hands, get upgraded or removed much more often then the  
proxy itself, which maintains its central role over time. Secondly,  
once you


That's why I prefer a "virtual" gateway which is hosted myself. The  
proxy does not send the calls directly to the gateway providers, but  
to the "virtual" gateway, which does LCR, accounting 


These virtual gateway can either be a B2BUA (in the simplest case  
Asterisk) or a SIP proxy with media relay or any other technique to  
make sure that the CDRs are correct.


Right, this is more or less what I had in mind. For the sake of  
simplicity I would do this without duplicating the proxy but this is  
just a detail. The key is to have something in the media path, having  
it you can always take the right decision, account for the right  
duration and terminate calls whenever is considered appropriate. Now,  
with the advanced capabilities of the dialog module I am not sure what  
more functionality related to accounting an external B2BUA can provide  
that cannot be provided by this tandem with the dialog module and the  
right server logic.


With the risk of stating the obvious here is what I mean:

http://cdrtool.ag-projects.com/attachment/wiki/WikiStart/OpenSIPS-accounting.png

Adrian



regards
klaus


do more the voice like video and other services that require  
billing and are not PSTN related, the SIP Proxy is the only network  
element that has access to the signalling and is able to generate  
accounting tickets.
Solving the accounting related problems at the SIP Proxy level is a  
worthwhile investment while other options are just temporary fixes  
that work in a particular case for a limited amount of time and  
that is a waste of money.

Adrian
On Jan 7, 2009, at 2:25 AM, Jiri Kuthan wrote:
authentication does not provide actually value here. dialog would  
not

either, since
the same trick can be achieved for example by low max-forwards.  
IMO the

proper
choice is accounting from the gateway, which provides the actual  
service.
A proxy can only provide an approximation which is inherentely to  
some

extent
more error-prone than the box doing the actual job.

-jiri

Bogdan-Andrei Iancu wrote:

Hi Iñaki,

Have you consider requesting auth for the BYE ? from SIP point of  
view

is perfectly valid

Regards,
Bogdan

Iñaki Baz Castillo wrote:
Hi, I'm thinking in the following flow in which the caller/ 
attacker

would get an unlimited call (but a limited CDR duration):

--
attacker OpenSIPS (Acc) 
gateway


INVITE (CSeq 12)  -->
< 407 Proxy Auth

INVITE (CSeq 13)  -->
INVITE (CSeq 13)   
-->
<---  
200 Ok

<--- 200 Ok
<< Acc START >>
ACK (CSeq 13) --->
ACK (CSeq 13)  
--->


<*** RTP >

# Fraudulent BYE !!!
BYE (CSeq 10) --->
<< Acc STOP >>
BYE (CSeq 10)  
--->
<-- 500 Req Out of  
Order

<-- 500 Req Out of Order
--

The call hasn't finished, but OpenSIPS has ended the accounting  
for

this call since it received a BYE. And this BYE will generate a
correct ACC Stop action (since it matches From_tag, To_tag and
Call-ID).

I think this is *VERY* dangerous and I hope I'm wrong.

Would help the dialog module here? does the dialog module check  
the

CSeq of the BYE in some way and could it prevent OpenSIPS from
generating the ACC STOP action? (I don't think so).

Any idea?








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Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

[Adrian Georgescu]

What exactly do you mean, can you re-phase the question?

Adrian


On Jan 7, 2009, at 2:17 PM, Jun.Wen wrote:

I've created an user in another opensips server and confirmed that  
user in the table of subscriber of opensips database. How can I let  
OpenXCAP to visit the mysql of OpenSIPS ?


The following is part of my OpenXCAP config.ini :
-
[Server]
address = 0.0.0.0
port = 80
root = http://xcap.mylab.net/xcap-root
backend = OpenSER
[Database]
authentication_db_uri = mysql://root:openx...@localhost/openxcap
storage_db_uri = mysql://root:openx...@localhost/openxcap
subscriber_table = subscriber
xcap_table = xcap

[OpenSER]
xmlrpc_url = http://osips.mylab.net:8080
-


From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月7日 15:49
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

The database is commonly shared between the two servers. So you can  
create an user with opensipsctl command on the sip machine as long  
as OpenXCAP is using the same database as OpenSIPS.


Adrian

On Jan 7, 2009, at 8:29 AM, Jun.Wen wrote:

I did not install opensips with openxcap in same server, is it a  
must or not ?


From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月7日 14:49
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

Did you try opensipsctl command?

Adrian

On Jan 7, 2009, at 5:33 AM, Jun.Wen wrote:


Adrian,

I guess my problem is I did not successfully created users in xcap  
server by the scripts of "python add-openxcap-user.py". Inside the  
table of subscriber of mysql, there is no any user record existed.


Any tips to create user inside xcap server ?

Regards

Jun

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月3日 1:08
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

I do not find in your logs what is wrong with your setup, it could  
be either the client or the server configuration.


Can you try your XCAP client using a SIP account from http://sip2sip.info 
 see if that works?


Regards,
Adrian


On Jan 2, 2009, at 2:39 PM, Jun.Wen wrote:


It looks like this -

-
xcap:~/.sipclient# cat config.ini
# rename this file to 'config.ini' and copy it in ~/.sipclient/  
directory


# this will be the default account used by xcapclient
[Account]
sip_address = al...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

# this will be used when -a bob command-line switch is provided
[Account_bob]
sip_address = b...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月2日 21:35
To: Jun.Wen
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What is the content of your xcapclient configuration  
file .sipclient/config.ini ?


Adrian

On Jan 2, 2009, at 9:33 AM, Jun.Wen wrote:


Hi, All, Happy new year to all the team here.

Please help me to figure out what is going wrong with my  
openxcap. Thanks in advance.


I've built up openxcap in debian according to the installation  
guide and the it works as followings info from error.log.

---
2009-01-02 02:56:26-0500 [-] Starting OpenXCAP 1.0.6
2009-01-02 02:56:37-0500 [-] Supported Root URIs: http://192.168.10.10/xcap-root
2009-01-02 02:56:39-0500 [-] Certificate file 'tls/server.crt'  
could not be loaded: File 'tls/server.crt' does not exist
2009-01-02 02:56:39-0500 [-] Private key file 'tls/server.key'  
could not be loaded: File 'tls/server.key' does not exist
2009-01-02 02:56:39-0500 [-] Trusted peers: 192.168.10.0/24,  
127.0.0.1
2009-01-02 02:56:39-0500 [-] xcap.server.HTTPFactory starting on  
80



The openxcap server is configured to domain 192.168.10.10  
( local server static IP address ). I also created a test  
account al...@192.168.10.10 with pwd 123 by python add-openxcap- 
user.py in scripts.  While when I tried the python-xcapclient to  
put a document to the openxcap server, I always encountered "401  
Unauthorized" for following details -

---
xcap:/opt/python-xcaplib/examples# xcapclient -i pres-rules.xml  
put

put 
http://192.168.10.10/xcap-root/pres-rules/users/sip:al...@192.168.10.10/index
401 Unauthorized
content-type: text/html
content-length: 1

Re: [OpenSIPS-Users] Upgrade CDRTool to 6.7.1 - problems

[Adrian Georgescu]

Yes, the differences in mysql schema between different versions are  
logged here:


http://cdrtool.ag-projects.com/browser/setup/mysql/alter_tables.mysql

Regards,
Adrian

On Jan 8, 2009, at 11:14 AM, Carlo Dimaggio wrote:



Il giorno 29/dic/08, alle ore 15:51, Adrian Georgescu ha scritto:


Carlo,

From your description it looks like an incorrect configuration in
global.inc.

If I were you I would start with a fresh  global.inc.simple.sample
and port the settings from the old version.

Adrian


Hi Adrian,

Thank you for your reply.
I created another global.inc starting from global.inc.simple.sample
but the problem still remained.
After that, I have dropped the cdrtool db and created another through
the new script setup_mysql.sh. (ver. 6.7.1)

Now the problem is solved and I can connect to the web interface. So I
think that there was some differences between the database schemas...


Regards,
Carlo

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Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

[Adrian Georgescu]

By setting:

authentication_db_uri = mysql://root:openx...@localhost/openxcap
storage_db_uri = mysql://root:openx...@localhost/openxcap

To your opensips database.

Adrian

On Jan 9, 2009, at 5:13 AM, wenjun wrote:


Adrian,

Sorry for my awkward question. I mean I've created a test user  
inside another machine of OpenSIPS. How can I configure OpenXCAP  
using the same database as OpenSIPS. I posted my config.ini of  
OpenXCAP here if you can see the that is correct or now.


Regards

Jun



CC: users@lists.opensips.org
From: a...@ag-projects.com
To: jun@msn.com
Date: Wed, 7 Jan 2009 16:42:51 +0100
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What exactly do you mean, can you re-phase the question?

Adrian


On Jan 7, 2009, at 2:17 PM, Jun.Wen wrote:

I've created an user in another opensips server and confirmed that  
user in the table of subscriber of opensips database. How can I let  
OpenXCAP to visit the mysql of OpenSIPS ?


The following is part of my OpenXCAP config.ini :
-
[Server]
address = 0.0.0.0
port = 80
root = http://xcap.mylab.net/xcap-root
backend = OpenSER
[Database]
authentication_db_uri = mysql://root:openx...@localhost/openxcap
storage_db_uri = mysql://root:openx...@localhost/openxcap
subscriber_table = subscriber
xcap_table = xcap

[OpenSER]
xmlrpc_url = http://osips.mylab.net:8080
-


From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月7日 15:49
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

The database is commonly shared between the two servers. So you can  
create an user with opensipsctl command on the sip machine as long  
as OpenXCAP is using the same database as OpenSIPS.


Adrian

On Jan 7, 2009, at 8:29 AM, Jun.Wen wrote:

I did not install opensips with openxcap in same server, is it a  
must or not ?


From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月7日 14:49
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

Did you try opensipsctl command?

Adrian

On Jan 7, 2009, at 5:33 AM, Jun.Wen wrote:

Adrian,

I guess my problem is I did not successfully created users in xcap  
server by the scripts of "python add-openxcap-user.py". Inside the  
table of subscriber of mysql, there is no any user record existed.


Any tips to create user inside xcap server ?

Regards

Jun

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月3日 1:08
To: Jun.Wen
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

I do not find in your logs what is wrong with your setup, it could  
be either the client or the server configuration.


Can you try your XCAP client using a SIP account fromhttp://sip2sip.info 
 see if that works?


Regards,
Adrian


On Jan 2, 2009, at 2:39 PM, Jun.Wen wrote:

It looks like this -

-
xcap:~/.sipclient# cat config.ini
# rename this file to 'config.ini' and copy it in ~/.sipclient/  
directory


# this will be the default account used by xcapclient
[Account]
sip_address = al...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

# this will be used when -a bob command-line switch is provided
[Account_bob]
sip_address = b...@192.168.10.10
password = 123
xcap_root = http://192.168.10.10/xcap-root

From: Adrian Georgescu [mailto:a...@ag-projects.com]
Sent: 2009年1月2日 21:35
To: Jun.Wen
Subject: Re: [OpenSIPS-Users] OpenXCAP Put 401 Unauthorized

What is the content of your xcapclient configuration file .sipclient/ 
config.ini ?


Adrian

On Jan 2, 2009, at 9:33 AM, Jun.Wen wrote:

Hi, All, Happy new year to all the team here.

Please help me to figure out what is going wrong with my openxcap.  
Thanks in advance.


I've built up openxcap in debian according to the installation guide  
and the it works as followings info from error.log.

---
2009-01-02 02:56:26-0500 [-] Starting OpenXCAP 1.0.6
2009-01-02 02:56:37-0500 [-] Supported Root URIs:http://192.168.10.10/xcap-root
2009-01-02 02:56:39-0500 [-] Certificate file 'tls/server.crt' could  
not be loaded: File 'tls/server.crt' does not exist
2009-01-02 02:56:39-0500 [-] Private key file 'tls/server.key' could  
not be loaded: File 'tls/server.key' does not exist

2009-01-02 02:56:39-0500 [-] Trusted peers: 192.168.10.0/24, 127.0.0.1
2009-01-02 02:56:39-0500 [-] xcap.server.HTTPFactory starting on 80


The openxcap server is configured to domain 192.168.10.10 ( local  
s

Re: [OpenSIPS-Users] radius SQL accounting of failed calls

[Adrian Georgescu]

On Jan 9, 2009, at 11:14 AM, Bogdan-Andrei Iancu wrote:

>> Thankfully, the patch buried within the contrib directory of  
>> CDRTool applied
>> well against freeradius-server-2.1.3.  Failed calls from OpenSIPS  
>> now cause
>> SQL records to be inserted.  Inspecting the inserted records,  
>> however, I
>> don't see anything indicating whether it was a START, STOP, or  
>> FAILED at the
>> radius level.  Perhaps the only indicator is the session time?  I'm  
>> still
>>
>> investigating.

The radius ticket in /var/log/fraaradius/radacct/* shows what type of  
packet it is.

Then in sql.conf you can configure the server to write into mysql per  
packet type what you need.

Adrian


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Re: [OpenSIPS-Users] radius SQL accounting of failed calls

[Adrian Georgescu]

Bogdan

The patch is here:

http://download.ag-projects.com/CDRTool/contrib/

Adrian


On Jan 9, 2009, at 11:14 AM, Bogdan-Andrei Iancu wrote:


Hi Jeff,

Could you point me the FREERADIUS patch you are talking about? just to
take a look and maybe push the discussion to a developer from Free  
Radius.


Thanks and regards,
Bogdan

Jeff Pyle wrote:

Hi Bogdan,

That makes sense to me.  In fact, that seemed to be the central  
point of the
argument against using a STOP record for a failed call:  "You can't  
STOP
what never STARTed in the first place."  Perhaps the argument from  
the other
side is that one must take a unique identifier into consideration?   
I cannot
claim to understand the implications yet.  I started playing with  
radius

accounting only last week.  :)

Thankfully, the patch buried within the contrib directory of  
CDRTool applied
well against freeradius-server-2.1.3.  Failed calls from OpenSIPS  
now cause
SQL records to be inserted.  Inspecting the inserted records,  
however, I
don't see anything indicating whether it was a START, STOP, or  
FAILED at the
radius level.  Perhaps the only indicator is the session time?  I'm  
still

investigating.


- Jeff





On 12/15/08 6:36 AM, "Bogdan-Andrei Iancu"   
wrote:




Hi Jeff,

OpenSIPS is still sending the FAILED values for the missed calls.  
From

ACC point of view, you have two cases - A) established calls
(START+STOP) and B) failed calls (FAILED).

If you use the STOP also for the failed calls, wouldn't be a  
confusion
in between the STOP of an ongoing call and the STOP of a failed  
call?


Regards,
Bogdan

Jeff Pyle wrote:


Hello,

It seems there was a long thread some time back about using the
so-called non-standard ³Acct-Status-Type = Failed² in the radius
packet. The two proposed solutions seemed to be 1) use a Stop type
instead of Failed, or 2) patch Freeradius.

Since I¹m still seeing the Failed type in the radius detail file,  
it

appears OpenSIPS didn¹t change to Stop.

So, anyone know of a patch for current Freeradius sources?


Thanks,
Jeff








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Re: [OpenSIPS-Users] [OpenSIPS-Devel] new CDRTool release 6.7.1

[Adrian Georgescu]

Default paths, configs, database names, support for MediaProxy2, sip  
trace module database schema and there could be others I lost track of.


Adrian

On Jan 9, 2009, at 10:09 AM, Klaus Darilion wrote:




Adrian Georgescu schrieb:

Hello,
The software contains improvements for compatibility with the  
latest  OpenSIPS and MediaProxy versions and the installation  
documentation.  An installation web page has been created at:

http://cdrtool.ag-projects.com/wiki/Install
Changelog
cdrtool (6.7.1) unstable; urgency=low
  * Simplified the sample configuration file global.inc.simple.sample
  * Improved installation documentation
cdrtool (6.7.0) unstable; urgency=low
  * Migrated to OpenSIPS, this version is incompatible with  
OpenSER,  do not

upgrade to this version if you still use OpenSER


Hi Adrian!

What exactly is the reason that it is incompatible with OpenSER? (I  
try to find out if it is incompatible to Kamailio too)


regards
klaus




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[OpenSIPS-Users] group_radius , radiusclient problem

[Adrian Georgescu]

The interfaces and backend used by OpenSIPS for functions like DNS,  
SQL and Radius impose much more relevant limits in scalability then  
OpenSIPS modules themselves.

 From your example, in my experience the limitation is most likely  
located in the Radius server rather than the client.

Did you check the server if it can handle the amount of requests sent  
by OpenSIPS?

Adrian


 >>>
I have tried group_radius on system with 35K users.
Check is being done for REGISTER and INVITE mgs. After enabling check
with gflag, registered cound falls by 10K.
In logs message below start appearing.

Jan  9 02:18:46 aster20 /usr/sbin/openser[31418]: rc_get_seqnbr:
couldn't get lock after 10 tries: /var/run/radius.seq
Jan  9 02:18:46 aster20 /usr/sbin/openser[31395]: rc_get_seqnbr:
couldn't get lock after 10 tries: /var/run/radius.seq
Jan  9 02:18:46 aster20 /usr/sbin/openser[31417]: rc_get_seqnbr:
couldn't get lock after 10 tries: /var/run/radius.seq

Looks like it's readiusclient problem. Has anyone experience something  
similar?


openser -V
version: openser 1.2.2-notls (i386/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: unknown
@(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $
main.c compiled on 01:12:52 Nov  5 2007 with gcc 4.1.2

-- 
Piotr Sobolewski
sobolewski at gmail.com



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Re: [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver

[Adrian Georgescu]

Ali,

Use UDP transport, it works reliable.

Adrian


On Jan 12, 2009, at 6:26 PM, Ali Jawad wrote:


But there are roughly only 100 users ...this is a quadro core server.



Fax: +961 1 375554


-Original Message-
From: Klaus Darilion [mailto:klaus.mailingli...@pernau.at]
Sent: 2009-01-12 15:40
To: Ali Jawad
Cc: users@lists.opensips.org; us...@lists.kamailio.org
Subject: Re: [OpenSIPS-Users] WARNING:core:send2child: no free tcp
receiver

Probably the proxy tries to open new tcp connections. This will fail
after some timeout. During waiting for the timeout the TCP processes  
are

busy, thus they can not handle new requests (as the warning says)

klaus

Ali Jawad schrieb:

Adding to the below


I am getting these errors

Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]:
ERROR:core:tcpconn_connect: tcp_blocking_connect failed Jan 12
12:35:30 sero /usr/local/sbin/openser[30076]:
ERROR:core:tcp_send: connect failed
Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]:

ERROR:sl:msg_send:

tcp_send failed
Jan 12 12:35:30 sero /usr/local/sbin/openser[30076]:
ERROR:auth:challenge: failed to send the response Jan 12 12:35:31  
sero



/usr/local/sbin/openser[30076]:
ERROR:core:tcp_blocking_connect: poll error: flags 18 Jan 12 12:35:31
sero /usr/local/sbin/openser[30076]:
ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111)
Connection refused


With Regards



Ali Jawad

System Administrator

Splendor Telecom (www.splendor.net )

Beirut, Lebanon

Phone: +961 1 373725

Fax: + 961 1 375554





--
--
*From:* Ali Jawad
*Sent:* 2009-01-12 12:33
*To:* Ali Jawad; users@lists.opensips.org
*Cc:* us...@lists.kamailio.org
*Subject:* RE: [OpenSIPS-Users] WARNING:core:send2child: no free tcp
receiver

As an update I have retrieved this


[r...@sero ~]# /usr/local/sbin/openserctl ps

Process:: ID=0 PID=30033 Type=attendant

Process:: ID=1 PID=30036 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=2 PID=30037 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=3 PID=30038 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=4 PID=30039 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=5 PID=30041 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=6 PID=30042 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=7 PID=30043 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=8 PID=30044 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=9 PID=30045 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=10 PID=30046 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=11 PID=30047 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=12 PID=30048 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=13 PID=30049 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=14 PID=30050 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=15 PID=30051 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=16 PID=30052 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=17 PID=30053 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=18 PID=30054 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=19 PID=30055 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=20 PID=30056 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=21 PID=30057 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=22 PID=30060 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=23 PID=30062 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=24 PID=30064 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=25 PID=30065 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=26 PID=30066 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=27 PID=30067 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=28 PID=30068 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=29 PID=30069 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=30 PID=30070 Type=SIP receiver udp:xx.yy.zz.aa:5060

Process:: ID=31 PID=30071 Type=timer

Process:: ID=32 PID=30072 Type=timer

Process:: ID=33 PID=30073 Type=MI FIFO

Process:: ID=34 PID=30074 Type=TCP receiver

Process:: ID=35 PID=30075 Type=TCP receiver

Process:: ID=36 PID=30076 Type=TCP receiver

Process:: ID=37 PID=30077 Type=TCP receiver

Process:: ID=38 PID=30079 Type=TCP receiver

Process:: ID=39 PID=30080 Type=TCP receiver

Process:: ID=40 PID=30081 Type=TCP receiver

Process:: ID=41 PID=30082 Type=TCP receiver

Process:: ID=42 PID=30084 Type=TCP receiver

Process:: ID=43 PID=30085 Type=TCP receiver

Process:: ID=44 PID=30086 Type=TCP receiver

Process:: ID=45 PID=30087 Type=TCP receiver

Process:: ID=46 PID=30090 Type=TCP receiver

Process:: ID=47 PID=30091 Type=TCP receiver

Process:: ID=48 PID=30092 Type=TCP receiver

Process:: ID=49 PID=30094 Type=TCP receiver

Process:: ID=50 PID=30095 Type=TCP receiver

Process:: ID=51 PID=30096 Type=TCP receiver

Process:: ID=52 PID=30097 Type=TCP receiver

Process:: ID=53 PID=30098 Type=TCP receiver

Process:: ID=54 PID=30099 Type=TCP receiver

Process:: ID=55 PID=30100 Type=TCP rece

Re: [OpenSIPS-Users] WARNING:core:send2child: no free tcp receiver

[Adrian Georgescu]

This is why we talk about new design. Exactly for this reason.

Just disable TCP in the server configuration, no xSER variant can work  
reliable with TCP today because of the blocking design.


Adrian


On Jan 12, 2009, at 6:50 PM, Iñaki Baz Castillo wrote:


2009/1/12 Adrian Georgescu :

Use UDP transport, it works reliable.


Ops, does it mean that UDP is more suitable in a SIP proxy/server  
than TCP?

I really would like SIP to migrate to TCP asap.

So, if OpenSIPS tries to do a TCP connection and it takes some time
(some seconds) until a timeout occurs, then that process is unable to
handle other SIP requests. an attacker could send just 100 SIP request
with ";transport=TCP" in the RURI and a RURI host which drops the TCP
connections.
This would cause all the OpenSIPS TCP processes being blocked !! is  
it?



--
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Re: [OpenSIPS-Users] jitter buffer problems

[Adrian Georgescu]

No, jitter is an end-point problem.

Adrian

On Jan 12, 2009, at 7:57 PM, Sergio Grimi wrote:


Hello,

I have an access gateway with jitter buffer problems. Is possible to  
adapt the media proxy or some other module of the opensips that  
manipulates streams of RTP/RTCP to solve this problem?


Is a viable solution to adapt the mediaproxy? There is some  
alternative that I am not considering?


Regards
Sergio E. Grimi
Ingeniería de Software
Tel: +54-341-4230504
Cel: +54-341-5325609
MSLC
sergio.gr...@mslc.com.ar
www.mslc.com.ar
Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica
Ciudad Universitaria Rosario UNR, CCT CONICET
Rosario - Santa Fé – Argentina

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Re: [OpenSIPS-Users] CDRTool CDR search display columns

[Adrian Georgescu]

This is obviously a bug. Can you tell me what Permissions you have set  
for the account you login with and what settings you have in  
global.inc for your OpenSIPS data source?


Adrian

On Jan 12, 2009, at 8:42 PM, Jeff Pyle wrote:


Hello,

When we search for CDRs in CDRTool 6.7.1, it appears the Price  
column is being skipped.  We have a correct value for Dur, but the  
KBIn value shows up in the Price column, KBOut under KBIn, Status  
shows up under KBOut, Codecs under Status, and the Codec column is  
empty.  Expanding the entry shows “Free call” under rating  
information.  That makes sense as have no rating information  
configured.


How might one get the display to have a blank column for Price, or  
eliminate it altogether?



Regards,
Jeff
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Re: [OpenSIPS-Users] CDRTool CDR search display columns

[Adrian Georgescu]

Hi Jeff,

Can you be more specific and tell me exactly what Permissions have you  
set in the web interface. There is nothing called Administrator.


Adrian


On Jan 12, 2009, at 9:02 PM, Jeff Pyle wrote:


Adrian,

Administrator privileges, and:

"opensips_radius"=>array(
"name"   => "OpenSIPS_fp1",
"class"  => "CDRS_opensips",
"db_class"   => array("DB_radius"),
"table"  => "radacct".date("Ym"),
"rateField"  => "Rate",
"rating" => "0",
"E164_class" => "E164_US",  // define a  
custom class to determine the E164 for a telephone number
// see  
E164 classes as example in library/cdr_generic.php

"priceField" => "Price",
"DestinationIdField" => "DestinationId",
"normalizedField"=> "Normalized",
"BillingPartyIdField"=> "UserName",
"db_susbcribers" => "DB_opensips",
"domain_table"   => "domain", // table  
of db_susbcribers that holds domains served by the sip proxy
"subscriber_table"   => "subscriber", // table  
of db_susbcribers that holds susbcribers served by the sip proxy
 // "intAccessCode"  => "011",//  
international acess code, numbers prefixed with this are considered  
international destinations
 // "natAccessCode"  => "1",  //  
international acess code, numbers prefixed with this are considered  
national destinations

"sipTrace"   => "sip_trace",
"mediaTrace" => "media_trace",
"UserQuotaClass" => "OpenSIPSQuota",
"UserQuotaNotify"=> "0",  //  
send e-mail notifications when quota is exceeded

"soapEngineId"   => '',
"domainTranslation"  => array(
   
"gw02.domain.com"  => "pstn.domain.com" // translate Realm

  ),
"SourceIPRealmTranslation"  => array(
  "10.0.0.1"  =>  
"gateway.example.com"  // translate Realm for sessions originating  
from IP address 10.0.0.1

  ),
"purgeCDRsAfter" => 999,   // how  
many days to keep old CDRs, valid only when Radius tables are not  
atomatically rotated
"db_registrar"   => "DB_opensips", //  
opensips location database
"enableThor" => false, // set to  
true if using SIP Thor
//"mediaSessions"  => "sipthor", //  
NGNPro engine id used by SIP Thor
//"networkStatus"  => "sipthor", //  
NGNPro engine id used by SIP Thor
"mediaDispatcher"=> "tcp:mediadispatcherhost: 
25061",// Where to get the active media sessions from MediaProxy  
2.0
   // Create 
 /etc/cdrtool/mediaproxy.pem containing the certificate and private  
key

"mediaServers"   => array(
  "0.0.0.0"
  )  // where to get  
the active sessions from MediaRroxy 1.x

),

- Jeff




On 1/12/09 2:50 PM, "Adrian Georgescu"  wrote:

This is obviously a bug. Can you tell me what Permissions you have  
set for the account you login with and what settings you have in  
global.inc for your OpenSIPS data source?


Adrian

On Jan 12, 2009, at 8:42 PM, Jeff Pyle wrote:


Hello,

 When we search for CDRs in CDRTool 6.7.1, it appears the Price  
column is being skipped.  We have a correct value for Dur, but the  
KBIn value shows up in the Price column, KBOut under KBIn, Status  
shows up under KBOut, Codecs under Status, and the Codec column is  
empty.  Expanding the entry shows “Free call” under rating  
information.  That makes sense as have no rating information  
configured.


 How might one get the display to have a blank column for Price,  
or eliminate it altogether?



 Regards,
 Jeff

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Re: [OpenSIPS-Users] CDRTool CDR search display columns

[Adrian Georgescu]

I tried with the default account and I cannot reproduce a missing  
column.


Could you paste the html code from your browser?

Adrian

On Jan 12, 2009, at 9:26 PM, Jeff Pyle wrote:


Adrian,

A bit green on my part perhaps but I’m not sure where to go in the  
web interface.  I’m using the “admin” user that comes pre-configured.




- Jeff



On 1/12/09 3:18 PM, "Adrian Georgescu"  wrote:


Hi Jeff,

Can you be more specific and tell me exactly what Permissions have  
you set in the web interface. There is nothing called Administrator.


Adrian


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Re: [OpenSIPS-Users] Integration of CDRtool(6.7.1) with openser 1.3.4

[Adrian Georgescu]

On Jan 13, 2009, at 8:28 AM, ASHWINI NAIDU wrote:


hi all,

 I have installed openser 1.3.4 and i am trying to integrate  
cdrtool 6.7.1 with it. As per i see it is compatible with opensips.


It is not.

If anybody has an idea abt what changes have to be made so that it  
works well with openser 1.3.4 please help.


--
Thankin You,
Ashwini BR Naidu
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Re: [OpenSIPS-Users] Integration of CDRtool(6.7.1) with openser 1.3.4

[Adrian Georgescu]

CDRTool 6.7.1 is not compatible with OpenSER. If you have previously  
installed CDRTool with OpenSER, do not upgrade top this version. The  
latest CDRTool version is compatible with OpenSIPS 1.4 or newer. See  
the changelog for more information.


Adrian


On Jan 13, 2009, at 11:57 AM, ASHWINI NAIDU wrote:


hi,

I didn't get you. can u say me what did you mena by saying "IT IS NOT"



On Tue, Jan 13, 2009 at 3:17 PM, Adrian Georgescu projects.com> wrote:


On Jan 13, 2009, at 8:28 AM, ASHWINI NAIDU wrote:


hi all,

 I have installed openser 1.3.4 and i am trying to integrate  
cdrtool 6.7.1 with it. As per i see it is compatible with opensips.


It is not.

If anybody has an idea abt what changes have to be made so that it  
works well with openser 1.3.4 please help.


--
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Ashwini BR Naidu
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Re: [OpenSIPS-Users] CDRTool CDR search display columns

[Adrian Georgescu]

I found the error and corrected in the trunk.

You can enable rating in global.inc for the moment, nothing dramatic  
will happen if you enable it except that the html output is correct.


Adrian

On Jan 12, 2009, at 9:26 PM, Jeff Pyle wrote:


Adrian,

A bit green on my part perhaps but I’m not sure where to go in the  
web interface.  I’m using the “admin” user that comes pre-configured.




- Jeff



On 1/12/09 3:18 PM, "Adrian Georgescu"  wrote:


Hi Jeff,

Can you be more specific and tell me exactly what Permissions have  
you set in the web interface. There is nothing called Administrator.


Adrian


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Re: [OpenSIPS-Users] Multi Tenant System

[Adrian Georgescu]

You can use OpenSIPS call_control module to terminate the calls on  
both side.


Adrian


On Jan 14, 2009, at 10:26 AM, ram wrote:




On 1/14/09, Mark Sayer  wrote:
I suggest using the pieces as they work best. Let OpenSIPs handle the
registration & NAT. Let Asterisk handle the media & connections to
terminators or PSTN. The only issue is that Asterisk will only handle
about 200 concurrent calls per box so a large installation might have
a single OpenSIPs box and multiple Asterisk boxes. Relatively simple
to setup and manage, stable, proven. Asterisk itself can be
"partitioned" through careful construction of the extensions.conf
file to do what you want.


Hi

thanks for the suggestion

thats what iam trying to achieve Asterisk (or freeswitch)

the suggestions again needed here

use Dispatcher Module or Drouting or LCR is again question

if its post paid fine

If its prepaid, how does that work of the Asterisk disconnect the call
how the Opensip react on the same

may be some are odd question, these all i have to understand


Ram



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Re: [OpenSIPS-Users] Multi Tenant System

[Adrian Georgescu]

No, you do not need Asterisk, what you need is depicted here:

http://callcontrol.ag-projects.com/

Adrian


On Jan 14, 2009, at 11:58 AM, ram wrote:




On 1/14/09, Adrian Georgescu  wrote:
You can use OpenSIPS call_control module to terminate the calls on  
both side.



Do i still need Asterisk if i use this module ?

ram




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Re: [OpenSIPS-Users] Multi Tenant System

[Adrian Georgescu]

The server is very stable is used for several years in production  
environments.


Adrian

On Jan 14, 2009, at 12:27 PM, ram wrote:




On 1/14/09, Adrian Georgescu  wrote:
No, you do not need Asterisk, what you need is depicted here:


http://callcontrol.ag-projects.com/

iam just reading the same

just want to know, how stable is this ?

any one using in production

Ram




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[OpenSIPS-Users] Digium Asterisk World, Feb 2-4, 2009 in Miami/Florida

[Adrian Georgescu]

Hello,

AG Projects will exhibit at ITEXPO on February 2-nd - 4-th, 2009 in  
Miami/Florida. We will provide presentations and live demonstrations  
with SIP SIMPLE presence with OpenSIPS and OpenXCAP and instant  
messaging (MSRP) with SIP SIMPLE client.

You can meet us at booth D13 in the Digium/Asterisk World area, if you  
want to abuse our time please drop me an email in private.

http://www.tmcnet.com/voip/conference/east-09/digium-asterisk-world.htm

Regards,
Adrian Georgescu


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Re: [OpenSIPS-Users] Integration of CDRtool(6.7.1) with openser 1.3.4

[Adrian Georgescu]

http://download.ag-projects.com/CDRTool/old/


On Jan 15, 2009, at 12:23 PM, ASHWINI NAIDU wrote:

where can i get the older versions of the CDR tool. I searched for  
it but i didn't see any. if u have the link to the repository please  
give it to me.


On Tue, Jan 13, 2009 at 9:44 PM, Adrian Georgescu projects.com> wrote:
CDRTool 6.7.1 is not compatible with OpenSER. If you have previously  
installed CDRTool with OpenSER, do not upgrade top this version. The  
latest CDRTool version is compatible with OpenSIPS 1.4 or newer. See  
the changelog for more information.


Adrian


On Jan 13, 2009, at 11:57 AM, ASHWINI NAIDU wrote:


hi,

I didn't get you. can u say me what did you mena by saying "IT IS  
NOT"




On Tue, Jan 13, 2009 at 3:17 PM, Adrian Georgescu projects.com> wrote:


On Jan 13, 2009, at 8:28 AM, ASHWINI NAIDU wrote:


hi all,

 I have installed openser 1.3.4 and i am trying to integrate  
cdrtool 6.7.1 with it. As per i see it is compatible with opensips.


It is not.

If anybody has an idea abt what changes have to be made so that it  
works well with openser 1.3.4 please help.


--
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Ashwini BR Naidu
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Ashwini BR Naidu


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Re: [OpenSIPS-Users] Problem on load mediaproxy

[Adrian Georgescu]

What you have pasted does not create any TLS certificates for  
MediaProxy, it simply copies them to CDRTool.


TLS certificate generation is general knowledge you can Google for it,  
there are many resources for how to create a TLS certificates.


Adrian

On Jan 13, 2009, at 12:21 PM, Alexandre Keller wrote:


Hi Mr. Klaver and list.

I've used the following how-to to create the certificates:

"
http://cdrtool.ag-projects.com/wiki/Install
Create or copy both a media relay certificate and its key to /etc/ 
cdrtool/mediaproxy.hostname.pem:


cat /etc/mediaproxy/tls/relay.crt >> /etc/cdrtool/ 
mediaproxy.hostname.pem
cat /etc/mediaproxy/tls/relay.key >> /etc/cdrtool/ 
mediaproxy.hostname.pem

"

Is it correct? Can you point me any other source of help?

Thanks in advance again.

--
Atenciosamente,

ALEXANDRE KELLER




+55 11 4063 - 9374
http://www.asteriks.com.br/

"Dinheiro é a consequência de um trabalho bem feito
e não o motivo para se fazer um bom trabalho."




On 13/01/2009, at 08:31, Ruud Klaver wrote:


Hi,

On 08 Jan 2009, at 14:52, Alexandre Keller wrote:


Hi there.

Anyone, please, can help me?

When I start mediaproxy the following messages appear.

Jan  8 11:29:59 sip media-relay[2292]: [-] Log opened.
Jan  8 11:29:59 sip media-relay[2292]: [-] Starting MediaProxy  
Relay 2.3.1
Jan  8 11:30:00 sip media-relay[2292]: [-] Set resource limit for  
maximum open file descriptors to 11000
Jan  8 11:30:00 sip media-relay[2292]: [-] fatal error: failed to  
create MediaProxy Relay: Base64 unexpected header error.
Jan  8 11:30:00 sip media-relay[2292]: [-] Traceback (most recent  
call last):
Jan  8 11:30:00 sip media-relay[2292]: [-] --- caught here> ---
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/usr/bin/ 
media-relay", line 54, in 
Jan  8 11:30:00 sip media-relay[2292]: [-] relay =  
MediaRelay()
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/mediaproxy/relay.py", line 295, in __init__
Jan  8 11:30:00 sip media-relay[2292]: [-] self.cred =  
X509Credentials(cert_name='relay')
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/mediaproxy/tls.py", line 132, in __init__
Jan  8 11:30:00 sip media-relay[2292]: [-]  
twisted.X509Credentials.__init__(self, self.X509cert,  
self.X509key, [self.X509ca], [self.X509crl])
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/mediaproxy/tls.py", line 99, in __get__
Jan  8 11:30:00 sip media-relay[2292]: [-] return  
descriptor.get()
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/mediaproxy/tls.py", line 82, in get
Jan  8 11:30:00 sip media-relay[2292]: [-] self.object =  
self.klass(f.read())
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "",  
line 1, in __init__

Jan  8 11:30:00 sip media-relay[2292]: [-]
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/gnutls/validators.py", line 273, in  
check_args
Jan  8 11:30:00 sip media-relay[2292]: [-] return  
func(*func_args)
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/gnutls/crypto.py", line 240, in __init__
Jan  8 11:30:00 sip media-relay[2292]: [-]  
gnutls_x509_crl_import(self._c_object, byref(data), format)
Jan  8 11:30:00 sip media-relay[2292]: [-]   File "/var/lib/ 
python-support/python2.5/gnutls/library/errors.py", line 64, in  
_check_status
Jan  8 11:30:00 sip media-relay[2292]: [-] raise  
GNUTLSError(ErrorMessage(retcode))
Jan  8 11:30:00 sip media-relay[2292]: [-]  
gnutls.errors.GNUTLSError: Base64 unexpected header error.


Thanks in advance

--
Atenciosamente,

ALEXANDRE KELLER


There seems to be an error while reading one of the certificate  
files you specified. You should check if they are in the correct  
format.


Ruud Klaver
AG Projects



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Re: [OpenSIPS-Users] CDRtool

[Adrian Georgescu]

You may have already normalized the CDRs when you had the destinations  
in your tables. The normalization updates the CDR table with the  
calculated destination id. If you normalize from scratch and still get  
data like that you obviously have provisioned it somehow, it cannot  
invent those numbers by itself without provisioning.


About dealing with custom prefixes you must craft an E164 class that  
deals with your dialing plan and use it instead of the default  
E164_Europe or E164_US. The updated Install wiki page describes this  
in more detail.


Adrian

On Jan 16, 2009, at 1:19 PM, Brian Chamberlain wrote:


Hi All,

I am doing some testing to normalise my destinations in the way that
CDRtool expects them to be before it normalises them.. :)

I have to do this as some of my providers make me do silly things with
prefixes etc. I know CDRTool allows me to put in a prefix to strip but
I have an array of them that I need to do.

Anyway, question..

I am testing to make sure my numbers are correct so I import my radius
records into a new table and when I look at the table in CDRTool it
seem to be matching my sip destinations to names eg.:

+1415 (VS (San Francisco) 1415)

This is confusing as I have nothing in my destination table.. Does it
do some kind of external lookup? I did have information in the
destinations table which I purged some time ago, maybe its cached
somewhere..

Thanks,
Brian











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Re: [OpenSIPS-Users] CDRtool

[Adrian Georgescu]

What you describe are the sample destinations in Dutch that come with  
the source code in a csv file that you have most probably loaded into  
your system. If you really had  an empty destinations table as you  
claim, they will not show up during normalization.


Adrian


On Jan 16, 2009, at 2:47 PM, Brian Chamberlain wrote:


Hi Adrian,

Thanks for the reply. I just imported my 200901 CDR's into a 200805  
table. I then ran some scripts to make the start of the SIP  
Destinations palatable to CDRTool.


CDRTool normalises them: +4417* (Verenigd Koninkrijk 44).  
These are fresh unnormalized CDR's fresh from OpenSIPS. My  
destinations table is empty.


With regard to the wiki are you talking about: 
http://cdrtool.ag-projects.com/browser/doc/RATING.txt

Thanks,
Brian

On 16 Jan 2009, at 13:26, Adrian Georgescu wrote:

You may have already normalized the CDRs when you had the  
destinations in your tables. The normalization updates the CDR  
table with the calculated destination id. If you normalize from  
scratch and still get data like that you obviously have provisioned  
it somehow, it cannot invent those numbers by itself without  
provisioning.


About dealing with custom prefixes you must craft an E164 class  
that deals with your dialing plan and use it instead of the default  
E164_Europe or E164_US. The updated Install wiki page describes  
this in more detail.


Adrian

On Jan 16, 2009, at 1:19 PM, Brian Chamberlain wrote:


Hi All,

I am doing some testing to normalise my destinations in the way that
CDRtool expects them to be before it normalises them.. :)

I have to do this as some of my providers make me do silly things  
with
prefixes etc. I know CDRTool allows me to put in a prefix to strip  
but

I have an array of them that I need to do.

Anyway, question..

I am testing to make sure my numbers are correct so I import my  
radius

records into a new table and when I look at the table in CDRTool it
seem to be matching my sip destinations to names eg.:

+1415 (VS (San Francisco) 1415)

This is confusing as I have nothing in my destination table.. Does  
it

do some kind of external lookup? I did have information in the
destinations table which I purged some time ago, maybe its cached
somewhere..

Thanks,
Brian











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[OpenSIPS-Users] New MediaProxy release 2.3.2

[Adrian Georgescu]

Hello,

There is a new release of MediaProxy available, it contains various  
bug fixes. To upgrade your debian installation:


apt-get update
apt-get install mediaproxy-dispatcher mediaproxy-relay mediaproxy-web- 
sessions


Or download the tar file from:

http://download.ag-projects.com/MediaProxy/

The changelog since 2.3.0 is below:

mediaproxy (2.3.2) unstable; urgency=low

  * Improved exception handling for DNS SRV lookups
  * Fixed exception caused by removing the dispatcher twice on shutdown
  * Fixed bug with preferred relay being considered last instead of  
first


mediaproxy (2.3.1) unstable; urgency=low

  * Changed info column in media_sessions to mediumblob to avoid  
truncation
  * Added missing minimum version dependencies in the debian control  
file
  * Fixed reconnect logic between relay and dispatcher to handle the  
case

when a dispatcher is removed while the connection to it has failed
  * Handle exception caused by a race condition when accessing  
counters from

a ForwardingRule that has expired but has not yet been acknowledged


Kind regards,
Adrian Georgescu


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Re: [OpenSIPS-Users] CDRTool destinations /rates question

[Adrian Georgescu]

Hi Brian,

The logic of the rating first determines the destination then it  
searches for a price for it. So for every entry in destinations table  
you MUST have an entry in the rates table otherwise the price is zero.


The best practice is to maintain a central minium destination table  
common for all customers (add entries to it as it grows) and define  
custom rates for each of them. Also if you have lot of resellers you  
can create a main rating table and add only exceptions for the  
destinations particular to some of them.


Adrian


On Jan 20, 2009, at 3:56 PM, Brian Chamberlain wrote:


Hi All,

I am sending calls to a number of different sip providers.
I have rates & destinations from all of them. Some of the providers
have broken up the amount of destinations into 30,000 different codes.
I am trying to build the rates and destinations tables so it is easy
to maintain in the future.

Would I be best having a minimal set of destinations to cover each
country and my local countries/areas and having the rates being more
specific.

I suppose my questions are the folowing.

If I have a destination:

1 USA

and a rate for 1 USA .02
and a rate for 1617 USA (Boston)

and the customer dials Boston then looking at the logic, even though I
don't have a boston Destination CDRTool will still rate the call using
the rate for 1617

If the reverse was through and I had a destination 1617 for boston but
only a rate for 1 USA would CDRTool use the 1 rate even though it
found the destination for 1617 in the destinations table?

Thanks,
Brian


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Re: [OpenSIPS-Users] CDRTool destinations /rates question

[Adrian Georgescu]

If dest is 1 only rate for dest 1 is applied. There is no longest  
match performed for a dest column in a rate table entry.


If you want a rate for 1617, add it to the dest table too.

Adrian

On Jan 20, 2009, at 4:19 PM, Brian Chamberlain wrote:


Hi Adrian,

Thanks for the quick response. As I thought!

Can you just confirm that if I have 1 as a destination,1 as a rate  
and also 1617 as a rate and 1617 is the number dialled then  
according to the documentation the rating engine will find the 1  
destination but will do a longest match and find 1617 as the rating  
record or am I hoping for too much?


Regards,
Brian

On 20 Jan 2009, at 15:03, Adrian Georgescu wrote:


Hi Brian,

The logic of the rating first determines the destination then it  
searches for a price for it. So for every entry in destinations  
table you MUST have an entry in the rates table otherwise the price  
is zero.


The best practice is to maintain a central minium destination table  
common for all customers (add entries to it as it grows) and define  
custom rates for each of them. Also if you have lot of resellers  
you can create a main rating table and add only exceptions for the  
destinations particular to some of them.


Adrian


On Jan 20, 2009, at 3:56 PM, Brian Chamberlain wrote:


Hi All,

I am sending calls to a number of different sip providers.
I have rates & destinations from all of them. Some of the providers
have broken up the amount of destinations into 30,000 different  
codes.

I am trying to build the rates and destinations tables so it is easy
to maintain in the future.

Would I be best having a minimal set of destinations to cover each
country and my local countries/areas and having the rates being more
specific.

I suppose my questions are the folowing.

If I have a destination:

1 USA

and a rate for 1 USA .02
and a rate for 1617 USA (Boston)

and the customer dials Boston then looking at the logic, even  
though I
don't have a boston Destination CDRTool will still rate the call  
using

the rate for 1617

If the reverse was through and I had a destination 1617 for boston  
but

only a rate for 1 USA would CDRTool use the 1 rate even though it
found the destination for 1617 in the destinations table?

Thanks,
Brian


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Brian Chamberlain
Dot Net Solutions Ltd.
68 Parkwest Enterprise Centre,
Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521

FAX:
[+353]
1
6237029


mobile:
[+353]
86
3883003



web:
www.asterisk.ie

* Looking for the most advanced PBX available that can also save you  
a fortune in communication costs?  asterisk.ie *



e-mail disclaimer

This e-mail and any files transmitted with it are confidential and  
intended
solely for the use of the individual or entity to whom they are  
addressed.
If you are not the intended recipient, you are hereby notified that  
any use

or dissemination of this communication is strictly prohibited.

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Re: [OpenSIPS-Users] [Serusers] SEMS 1.1.0 released

[Adrian Georgescu]

Hello Stefan,

For my understanding, does SEMS depend on a certain type SER  
deployment or is independent of it? Is a technical question.


Regards,
Adrian

On Jan 20, 2009, at 9:52 PM, Stefan Sayer wrote:


Hello,

Version 1.1.0 of SEMS, the SIP Express Media Server, has been  
released. SEMS is a free, high performance, extensible media and  
application server for SIP (RFC3261) based VoIP services.


New in version 1.1 is:
 * DSM state machine scripting (it's cool!)
 * an (experimental) ISDN gateway module
 * binrpc: MT (SER->) and connection pool (->SER)
 * MT xmlrpc server
 * controlled server shutdown
 * improved logging
 * g722 in 8khz compat mode
 * out of dialog request handling for modules & dialogs without
   sessions
 * audio file autorewind, AmAudio mixing
 * SIP and media IP separately configurable
 * UID/DID support for voicemail/-box/annrecorder
 * and quite some bugs and mem leaks fixed, documentation, etc.

You can get the sources from
ftp://ftp.iptel.org/pub/sems/sems-1.1.0.tar.gz
and find documentation and packages at
ftp://ftp.iptel.org/pub/sems/1.1/1.1.0/
and a debian repository (etch/lenny) at
deb http://ftp.iptel.org/pub/sems/debian etch free
deb-src http://ftp.iptel.org/pub/sems/debian etch free

Further documentation, links to lists, tracker etc is available at  
the project's homepage: http://iptel.org/sems/


I would like to thank for all contributions and patches, this time  
from

 Stefan Sayer, Raphael Coeffic, Grzegorz Stanislawski,
 Bogdan Pintea, Greger Teigre, Rui Jin Zheng,
 Alfred E Heggestad, Juha Heinanen, Peter Lemenkov,
 Peter Loeppky, Robert Szokovacs, Jeremy A, Alex Gradinar

Apologies if you receive multiple copies of this message.

Best Regards
Stefan Sayer

--
Stefan Sayer
VoIP services

stefan.sa...@iptego.com
www.iptego.com

IPTEGO GmbH
Am Borsigturm 40
13507 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
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Re: [OpenSIPS-Users] CDRTool destinations /rates question

[Adrian Georgescu]

Alex, I am trying to understand what precisely you are trying to  
achieve. What precisely are you working around that cannot be done in  
a natural way?


Adrian

On Jan 20, 2009, at 7:29 PM, Alex Balashov wrote:

Good workaround is to use translations in the proxy to prepend a  
prefix for each carrier to the DNIS so you can set the rating engine  
loose on that.


This is how billing systems attached to traditional softswitch EMSs  
work.


Brian Chamberlain wrote:


Thanks Adrian,
As I said, just trying to find an efficient way of doing this, all  
the  providers use different destination names, some have codes  
that don't  exist in the other's databases so trying to pull it all  
together in  CDRtool is proving a bit testing.

It is mentioned as a known limitation
'The rating engine does not calculate prices based on the outbound   
carriers or outbound gateways, the rating plan is is assigned by  
the  calling party and not by called party.'
I guess I am trying to figure out an efficient way to deal with  
the  slight nuances with different providers destination codes and   
descriptions and the overlaps in between..
If it was possible to rate with the destination gateway it would  
make  things a lot easier.

Thanks,
Brian
On 20 Jan 2009, at 15:38, Adrian Georgescu wrote:
If dest is 1 only rate for dest 1 is applied. There is no longest   
match performed for a dest column in a rate table entry.


If you want a rate for 1617, add it to the dest table too.

Adrian

On Jan 20, 2009, at 4:19 PM, Brian Chamberlain wrote:


Hi Adrian,

Thanks for the quick response. As I thought!

Can you just confirm that if I have 1 as a destination,1 as a  
rate  and also 1617 as a rate and 1617 is the number dialled  
then  according to the documentation the rating engine will find  
the 1  destination but will do a longest match and find 1617 as  
the rating  record or am I hoping for too much?


Regards,
Brian

On 20 Jan 2009, at 15:03, Adrian Georgescu wrote:


Hi Brian,

The logic of the rating first determines the destination then  
it  searches for a price for it. So for every entry in  
destinations  table you MUST have an entry in the rates table  
otherwise the  price is zero.


The best practice is to maintain a central minium destination   
table common for all customers (add entries to it as it grows)  
and  define custom rates for each of them. Also if you have lot  
of  resellers you can create a main rating table and add only   
exceptions for the destinations particular to some of them.


Adrian


On Jan 20, 2009, at 3:56 PM, Brian Chamberlain wrote:


Hi All,

I am sending calls to a number of different sip providers.
I have rates & destinations from all of them. Some of the  
providers
have broken up the amount of destinations into 30,000  
different  codes.
I am trying to build the rates and destinations tables so it  
is  easy

to maintain in the future.

Would I be best having a minimal set of destinations to cover  
each
country and my local countries/areas and having the rates  
being  more

specific.

I suppose my questions are the folowing.

If I have a destination:

1 USA

and a rate for 1 USA .02
and a rate for 1617 USA (Boston)

and the customer dials Boston then looking at the logic, even   
though I
don't have a boston Destination CDRTool will still rate the  
call  using

the rate for 1617

If the reverse was through and I had a destination 1617 for   
boston but

only a rate for 1 USA would CDRTool use the 1 rate even though it
found the destination for 1617 in the destinations table?

Thanks,
Brian


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Brian Chamberlain
Dot Net Solutions Ltd.
68 Parkwest Enterprise Centre,
Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521

FAX:
[+353]
1
6237029


mobile:
[+353]
86
3883003



web:
www.asterisk.ie

* Looking for the most advanced PBX available that can also save   
you a fortune in communication costs?  asterisk.ie *



e-mail disclaimer

This e-mail and any files transmitted with it are confidential  
and  intended
solely for the use of the individual or entity to whom they are   
addressed.
If you are not the intended recipient, you are hereby notified  
that  any use

or dissemination of this communication is strictly prohibited.

If you have received this e-mail in error, please advise the sender
immediately, then delete this e-mail.








Brian Chamberlain
Dot Net Solutions Ltd.
68 Parkwest Enterprise Centre,
Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521
FAX:
[+353]
1
6237029
mobile:
[+353]
86
3883003
web:
www.asterisk.ie
* Looking for the most advanced PBX available that can also save  
you a  fortune in communication costs?  asterisk.ie *

e-mail disclaimer
This e-mail and any files transmitted with it are confidential and   
intended
solely for the use of the individual or entity to whom they

Re: [OpenSIPS-Users] CDRTool destinations /rates question

[Adrian Georgescu]

The format of the actual dialed number is a routing problem and a  
configuration of the proxy issue. In CDRTool you always end-up with a  
normalized E164 number regardless of how the request URI was formated  
for the actual terminating gateway.

You may use any translation rules and LCR without having to change or  
configure the rating engine for this.

Adrian


On Jan 21, 2009, at 12:50 AM, Alex Balashov wrote:

> Unlike Brian, I am not familiar with CDRtool beyond a cursory level,  
> so perhaps I'm headed down the wrong track here.
>
> The general problem seems to be that the multiple destination  
> problem (variable-length prefixes) is multidimensional, so it is not  
> just a matter of sending to the longest dial prefix match for a  
> given destination.  The carrier must also be taken into account.   
> So, what is needed seems to be a destination metric that is a  
> composite rate of a gateway and a longest-prefix destination.
>
> The terminating carriers are fixed by a static LCR process.
>
> Is that right, Brian?
>
> Adrian Georgescu wrote:
>
>>

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Re: [OpenSIPS-Users] CDRTool destinations /rates question

[Adrian Georgescu]

The format of the actual dialed number is a routing problem and a  
configuration of the proxy issue. In CDRTool you always end-up with a  
normalized E164 number regardless of how the request URI was formated  
for the actual terminating gateway.


You may use any translation rules and LCR without having to change or  
configure the rating engine for this.


Adrian


On Jan 21, 2009, at 12:50 AM, Alex Balashov wrote:

Unlike Brian, I am not familiar with CDRtool beyond a cursory level,  
so perhaps I'm headed down the wrong track here.


The general problem seems to be that the multiple destination  
problem (variable-length prefixes) is multidimensional, so it is not  
just a matter of sending to the longest dial prefix match for a  
given destination.  The carrier must also be taken into account.   
So, what is needed seems to be a destination metric that is a  
composite rate of a gateway and a longest-prefix destination.


The terminating carriers are fixed by a static LCR process.

Is that right, Brian?

Adrian Georgescu wrote:

Alex, I am trying to understand what precisely you are trying to  
achieve. What precisely are you working around that cannot be done  
in a natural way?

Adrian
On Jan 20, 2009, at 7:29 PM, Alex Balashov wrote:
Good workaround is to use translations in the proxy to prepend a  
prefix for each carrier to the DNIS so you can set the rating  
engine loose on that.


This is how billing systems attached to traditional softswitch  
EMSs work.


Brian Chamberlain wrote:


Thanks Adrian,
As I said, just trying to find an efficient way of doing this,  
all the  providers use different destination names, some have  
codes that don't  exist in the other's databases so trying to  
pull it all together in  CDRtool is proving a bit testing.

It is mentioned as a known limitation
'The rating engine does not calculate prices based on the  
outbound  carriers or outbound gateways, the rating plan is is  
assigned by the  calling party and not by called party.'
I guess I am trying to figure out an efficient way to deal with  
the  slight nuances with different providers destination codes  
and  descriptions and the overlaps in between..
If it was possible to rate with the destination gateway it would  
make  things a lot easier.

Thanks,
Brian
On 20 Jan 2009, at 15:38, Adrian Georgescu wrote:
If dest is 1 only rate for dest 1 is applied. There is no  
longest  match performed for a dest column in a rate table entry.


If you want a rate for 1617, add it to the dest table too.

Adrian

On Jan 20, 2009, at 4:19 PM, Brian Chamberlain wrote:


Hi Adrian,

Thanks for the quick response. As I thought!

Can you just confirm that if I have 1 as a destination,1 as a  
rate  and also 1617 as a rate and 1617 is the number dialled  
then  according to the documentation the rating engine will  
find the 1  destination but will do a longest match and find  
1617 as the rating  record or am I hoping for too much?


Regards,
Brian

On 20 Jan 2009, at 15:03, Adrian Georgescu wrote:


Hi Brian,

The logic of the rating first determines the destination then  
it  searches for a price for it. So for every entry in  
destinations  table you MUST have an entry in the rates table  
otherwise the  price is zero.


The best practice is to maintain a central minium destination   
table common for all customers (add entries to it as it grows)  
and  define custom rates for each of them. Also if you have  
lot of  resellers you can create a main rating table and add  
only  exceptions for the destinations particular to some of  
them.


Adrian


On Jan 20, 2009, at 3:56 PM, Brian Chamberlain wrote:


Hi All,

I am sending calls to a number of different sip providers.
I have rates & destinations from all of them. Some of the  
providers
have broken up the amount of destinations into 30,000  
different  codes.
I am trying to build the rates and destinations tables so it  
is  easy

to maintain in the future.

Would I be best having a minimal set of destinations to cover  
each
country and my local countries/areas and having the rates  
being  more

specific.

I suppose my questions are the folowing.

If I have a destination:

1 USA

and a rate for 1 USA .02
and a rate for 1617 USA (Boston)

and the customer dials Boston then looking at the logic,  
even  though I
don't have a boston Destination CDRTool will still rate the  
call  using

the rate for 1617

If the reverse was through and I had a destination 1617 for   
boston but
only a rate for 1 USA would CDRTool use the 1 rate even  
though it

found the destination for 1617 in the destinations table?

Thanks,
Brian


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Dot Net Solutions Ltd.
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Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521

F

Re: [OpenSIPS-Users] Contributing to cdrtool

[Adrian Georgescu]

Hi Joan,

See  answers inline,

On Jan 22, 2009, at 6:42 PM, Joan wrote:


Good afternoon,

I am trying to install cdrtool to an ubuntu 8.04, there are some
problems I found so far:

.- I'd prefer to build myself the .deb, so everything is on the right
place whithout having to tweak everything by hand, but I cannot find
the Makefile, makedeb, whatever, any doc explaining how to do it?


Use debuild command


.- I installed darcs client into my system. When doing the darcs get,
I get the following:
darcs failed:  Not a repository:
http://devel.ag-projects.com/repositories/cdrtool (Failed to download
URL http://devel.ag-projects.com/repositories/cdrtool/_darcs/ 
inventory:

HTTP response code said error)


Fixed now, try again.


.- Is the ticket system alive, there's only a single ticket nobody
seem to care.
Also I don't see the timeline part of the trac (where
you can see the commits, changes to wiki, etc...)?


http://cdrtool.ag-projects.com/timeline


.- Is there any repository similar to the svnview so I can see the
changes to the code through a web browser?


http://cdrtool.ag-projects.com/browser


.- Any place to contribute?


Feel free to propose something on this mailing list.


It seems to me that for the minimal
installation (no radius server, no prepaid service), it's quite
confusing having to enable everything


Without a Radius setup CDRTool is pretty much useless.

Adrian


Kind regards

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Re: [OpenSIPS-Users] Fix for the freeradius patch (within cdrtool) for version 1.1.7

[Adrian Georgescu]

Just to double check where did you get the latest sources from?

Adrian


On Jan 23, 2009, at 12:34 PM, Joan wrote:


When following the details on how to compile freeradius with to
support cdrtool from CDRTool/setup/radius/FreeRadius/readme.txt

The patch there applies cleanly, there's only a issue when applying
the patch I attached to the mail, because it is already fixed in 1.1.7
version. Other than that, the patch applies cleanly.

--- freeradius.orig/src/modules/rlm_sql/drivers/rlm_sql_mysql/ 
sql_mysql.c

 2005-12-09 17:10:08.0 +0100
+++ freeradius.new/src/modules/rlm_sql/drivers/rlm_sql_mysql/ 
sql_mysql.c

  2006-12-12 23:59:11.0 +0100
@@ -82,7 +82,7 @@
   config->sql_db,
   atoi(config- 
>sql_port),

   NULL,
-
CLIENT_FOUND_ROWS))) {

+
CLIENT_FOUND_ROWS|CLIENT_MULTI_RESULTS))) {
   radlog(L_ERR, "rlm_sql_mysql: Couldn't connect socket
to MySQL server %...@%s:%s", config->sql_login, config->sql_server,
config->sql_db);
   radlog(L_ERR, "rlm_sql_mysql: Mysql error '%s'",
mysql_error(&mysql_sock->conn));
   mysql_sock->sock = NULL;

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Re: [OpenSIPS-Users] Call recording

[Adrian Georgescu]

On Jan 25, 2009, at 12:12 PM, Jimmy Svensson wrote:


Hi!

I'm looking for a pure call recording solution that generates pcap- 
files of the data from one specific call. I've been looking at using  
the a merged OpenSIPS+rtpproxy solution to achieve this. The  
rtpproxy would not be proxying any RTP traffic in this case, just  
terminating and recording. Does anyone have experience with this  
kind of solution?


Now I found that rtpproxy seems to be replaced by mediaproxy but I  
can't find any call recording features on the mediaproxy. Does  
anyone know if there are any plans to implement call recording in  
mediaproxy or do I have to go for the rtpproxy?


No plans at this moment.

Adrian


Thanks!
/Jimmy
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Re: [OpenSIPS-Users] Fix for the freeradius patch (within cdrtool) for version 1.1.7

[Adrian Georgescu]

OK thanks


On Jan 26, 2009, at 10:49 AM, Joan wrote:


2009/1/23 Adrian Georgescu :

Just to double check where did you get the latest sources from?

I got the cdrtool code from your site, the lastest packet version.
.- wget http://download.ag-projects.com/CDRTool/cdrtool_6.7.2.tar.gz
I finally could checkout your code from trunk, and the
freeradius.patch file is the same than the one I've got in the tar.gz.

So it seems that between 1.1.3 and 1.1.7 your patch was applied to the
official freeradius sources.



On Jan 23, 2009, at 12:34 PM, Joan wrote:

When following the details on how to compile freeradius with to
support cdrtool from CDRTool/setup/radius/FreeRadius/readme.txt

The patch there applies cleanly, there's only a issue when applying
the patch I attached to the mail, because it is already fixed in  
1.1.7

version. Other than that, the patch applies cleanly.

--- freeradius.orig/src/modules/rlm_sql/drivers/rlm_sql_mysql/ 
sql_mysql.c

2005-12-09 17:10:08.0 +0100
+++ freeradius.new/src/modules/rlm_sql/drivers/rlm_sql_mysql/ 
sql_mysql.c

 2006-12-12 23:59:11.0 +0100
@@ -82,7 +82,7 @@
  config->sql_db,
  atoi(config- 
>sql_port),

  NULL,
-
CLIENT_FOUND_ROWS))) {

+
CLIENT_FOUND_ROWS|CLIENT_MULTI_RESULTS))) {
  radlog(L_ERR, "rlm_sql_mysql: Couldn't connect socket
to MySQL server %...@%s:%s", config->sql_login, config->sql_server,
config->sql_db);
  radlog(L_ERR, "rlm_sql_mysql: Mysql error '%s'",
mysql_error(&mysql_sock->conn));
  mysql_sock->sock = NULL;

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[OpenSIPS-Users] New OpenXCAP release 1.0.7

[Adrian Georgescu]

Hello,

There are is a new release of the open source OpenXCAP server available.

For Debian and Ubuntu on i386 and amd64 architectures there is an  
official

public repository provided by AG Projects.

Install the AG Projects debian software signing key:

wget http://download.ag-projects.com/agp-debian-gpg.key
apt-key add agp-debian-gpg.key

Add these lines to etc/apt/sources.list

# AG Projects software
deb http://ag-projects.com/debian unstable main
deb-src http://ag-projects.com/debian unstable main

After that, run:

apt-get update
apt-get install openxcap

For other  operating systems download the tar archive from:

http://openxcap.org

Changelog:

openxcap (1.0.7) unstable; urgency=low

  * Use pysupport instead of pycentral for debian packaging
  * Log errors to syslog unless log_error_to_file=yes in [Logging]  
section
  * Fixed parsing node selectors of type /*[1] and added tests for  
this case
  * Fixed "unbound prefix" bug and added a test case. A new function  
is added

that checks well-formedness of the elements.
  * Only import gnutls when TLS is used
  * Fixed "watchers" application to generate and insert ETag header  
in the

response
  * Added test for "watchers" application (test_watchers.py)
  * Removed ./debian/openxcap.postinst
  * Updated sipthor interface for SIPThor 1.0.1
  * Replace openser-mi-proxy with opensips-mi-proxy in documentation
  * Improved INSTALL and synced it with the wiki
  * Updated README and TODO
  * Corrected name of pres-rules auid
  * Test system: updated to use python-xcaplib 1.0.8
  * Test system: added -r, --repeat option
  * Test system: added support for "--client eventlet"
  * Test system: added support for "--start-server config_file"
  * Test system: fixed to use original options for each test
  * Test system: added support for starting and using in-process server
(requires eventlet)
  * Test system: added undocumented '--client xcapclient' option. It  
makes
test system to use xcapclient tool instead of xcaplib package.  
This is

only useful for testing xcapclient.
  * More detailed report for test_auth.py
  * Fixed typo in the http error response
  * Made xcap.uri module accept parameter of uri to parse when run as  
script

(for simple troubleshooting)

Kind regards,
Adrian Georgescu

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Re: [OpenSIPS-Users] CDRTool: How to disable quota completely?

[Adrian Georgescu]

Hi Ikar,

If your data source does not have UserQuotaClass defined, the scripts  
that check the quota are not performing any actions. This exclude the  
SELECT in the opensips.subscriber table so you do not need to add a  
quota column either.


Internally the cdrtool.quota_usage MySQL table is always updated  
during the the CDR normalization but this does not require any changes  
in OpenSIPS tables.


For my understanding, what do you want to gain by disabling these  
internal UPDATEs?


Regards,
Adrian

On Feb 1, 2009, at 1:21 PM, happy.neko wrote:


Hi,

I want to disable CDRTool's quota functionality completely including  
SELECTs with quota column from subscriber table, UPDATEs during cdrs  
normalization etc.

How can I do this?

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Re: [OpenSIPS-Users] CDRTool: How to disable quota completely?

[Adrian Georgescu]

Hi Happy,

I have disabled all these queries in the trunk version. If you do not  
want to wait for the next release you can download it from darcs.


I hope you are now as your name implies :-)

Adrian


On Feb 2, 2009, at 10:18 PM, happy.neko wrote:


Hi Adrian,

I don't have UserQuotaClass in my global.inc, but SELECTs are  
executed according to syslog:


Feb  2 15:55:02 cdr cdrtool[25649]: ConnectFee=0. callid=c2a5537d-1641a...@192.168.1.113 
 Span=1 Duration=60 DestId=790433 default Profile=500 Period=w
eekday Rate=default Interval=0-24 Cost=2.2400/60 Price=2.2400 Price  
in=1.6000
Feb  2 15:55:02 cdr cdrtool[25649]: Database error: Invalid SQL:  
select quota from subscriber where username = 'XX' and domain =  
YYY'

Feb  2 15:55:02 cdr cdrtool[25649]: 67

I want to disable quota because of performance concerns (my DB  
already is under heavy load with opensips and legacy applications)  
and a bit of paranoid thoughts about functionality I'm not fully  
understand and need.


On Sun, Feb 1, 2009 at 4:37 PM, Adrian Georgescu projects.com> wrote:

Hi Ikar,

If your data source does not have UserQuotaClass defined, the  
scripts that check the quota are not performing any actions. This  
exclude the SELECT in the opensips.subscriber table so you do not  
need to add a quota column either.


Internally the cdrtool.quota_usage MySQL table is always updated  
during the the CDR normalization but this does not require any  
changes in OpenSIPS tables.


For my understanding, what do you want to gain by disabling these  
internal UPDATEs?


Regards,
Adrian

On Feb 1, 2009, at 1:21 PM, happy.neko wrote:


Hi,

I want to disable CDRTool's quota functionality completely  
including SELECTs with quota column from subscriber table, UPDATEs  
during cdrs normalization etc.

How can I do this?

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Re: [OpenSIPS-Users] [NEW Module] SIP Identity

[Adrian Georgescu]

Beyond being plain interesting, it is the most cost-efective way to  
implement secure identity between SIP Proxies serving different domains.


Adrian

On Feb 10, 2009, at 8:57 PM, Iñaki Baz Castillo wrote:


El Martes, 10 de Febrero de 2009, Bogdan-Andrei Iancu escribió:

Hello,


OpenSIPS 1.5.0 has a new module. The "identity" module is an
implementation of SIP identity as per RFC 4474
(http://www.ietf.org/rfc/rfc4474.txt).

Abstract (from RFC) :

  The existing security mechanisms in the Session Initiation Protocol
  (SIP) are inadequate for cryptographically assuring the identity of
  the end users that originate SIP requests, especially in an
  interdomain context.  This document defines a mechanism for  
securely
  identifying originators of SIP messages.  It does so by defining  
two

  new SIP header fields, Identity, for conveying a signature used for
  validating the identity, and Identity-Info, for conveying a  
reference

  to the certificate of the signer


Really interesting :)


--
Iñaki Baz Castillo

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Re: [OpenSIPS-Users] Paid Consultation Request

[Adrian Georgescu]

Geoffrey,

In my experience nobody on this mailing list with enough knowledge in  
this matter will be able to help fix all your possible miss- 
configurations as it can span fixing the whole universe depending on  
an infinit matrix of possibilities.


A 2 hour 'blessing' is most likely exactly what you are going to get  
for 300$. If you are the one responsable for 'a very high profile  
network'  and hope that by a 2 hour quick check you will be out of  
trouble, think that this has a very low probability to happen.


So nobody good enough for the job will take your request seriously.

Adrian

On Feb 11, 2009, at 5:46 PM, Geoffrey Mina wrote:


Hello,
I am looking for anyone who would consider themselves an 'expert' in
the field of OpenSIPS.  My company is launching an OpenSIPS deployment
to front-end all the SIP traffic entering our network.  I would like
to have someone experienced look over my config to give it the
proverbial 'blessing'.

We run a very high profile network and I can't afford to have any
minor misconfigurations or problems cause issues down the road.

I would be willing to pay $150/hr (USD) via PayPal.  I am guessing I
will need 2(ish) hours, and IM/MSN/Skype chat would be the best.

If anyone is interested, please let me know.
Thanks,
Geoff

p.s. we are primarily using the dispatcher module, so any interested
party should know that one inside and out!

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Re: [OpenSIPS-Users] Paid Consultation Request

[Adrian Georgescu]

Please let us all know if you get the blessing at least we know we are  
wrong.


Adrian

On Feb 11, 2009, at 6:25 PM, Geoffrey Mina wrote:


Thanks for your reply, but I tend to disagree.  I have spent many
hours programming my OpenSIPs deployment as well as processing test
calls.  It is a VERY simple deployment, which is why I think that 2
hours will be more than enough.  The factors are:

1 - We have one SIP provider
2 - We are ONLY handling INVITE requests for the purpose of load
balancing IVR systems
3 - We are using dispatcher in round-robin mode

With the scope of functionality being so narrow, I see no reason why
someone with some good experience wouldn't be able to simply
double-check my work.  This is my first fore into OpenSIPs, but
certainly not my first into SIP.

Looking forward to some other responses.

Thanks,
Geoff

On Wed, Feb 11, 2009 at 12:06 PM, Adrian Georgescu projects.com> wrote:

Geoffrey,
In my experience nobody on this mailing list with enough knowledge  
in this
matter will be able to help fix all your possible miss- 
configurations as it

can span fixing the whole universe depending on an infinit matrix
of possibilities.
A 2 hour 'blessing' is most likely exactly what you are going to  
get for
300$. If you are the one responsable for 'a very high profile  
network'  and
hope that by a 2 hour quick check you will be out of trouble, think  
that

this has a very low probability to happen.
So nobody good enough for the job will take your request seriously.
Adrian
On Feb 11, 2009, at 5:46 PM, Geoffrey Mina wrote:

Hello,
I am looking for anyone who would consider themselves an 'expert' in
the field of OpenSIPS.  My company is launching an OpenSIPS  
deployment

to front-end all the SIP traffic entering our network.  I would like
to have someone experienced look over my config to give it the
proverbial 'blessing'.

We run a very high profile network and I can't afford to have any
minor misconfigurations or problems cause issues down the road.

I would be willing to pay $150/hr (USD) via PayPal.  I am guessing I
will need 2(ish) hours, and IM/MSN/Skype chat would be the best.

If anyone is interested, please let me know.
Thanks,
Geoff

p.s. we are primarily using the dispatcher module, so any interested
party should know that one inside and out!

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Re: [OpenSIPS-Users] Paid Consultation Request

[Adrian Georgescu]

Geoffrey,

I am trying to help you fine tune your expectations. Anyone can  
briefly tell you what is wrong as they spot some issue. But a good  
consultant would tell you what you have missed too and how to solve  
problems you did not foresee yet. So by asking the wrong question and  
setting up the expectation so low you will get the same kind of answer  
and quality back.


Adrian


On Feb 11, 2009, at 6:29 PM, Geoffrey Mina wrote:


If it is simply a matter of financials, I would be willing to discuss
that matter with any interested party off-line.

thanks.

On Wed, Feb 11, 2009 at 12:25 PM, Alex Balashov
 wrote:

Iñaki Baz Castillo wrote:

2009/2/11 Adrian Georgescu :

Geoffrey,
In my experience nobody on this mailing list with enough  
knowledge in this
matter will be able to help fix all your possible miss- 
configurations as it

can span fixing the whole universe depending on an infinit matrix
of possibilities.
A 2 hour 'blessing' is most likely exactly what you are going to  
get for
300$. If you are the one responsable for 'a very high profile  
network'  and
hope that by a 2 hour quick check you will be out of trouble,  
think that

this has a very low probability to happen.
So nobody good enough for the job will take your request seriously.


I fully agree. Nobody, including the most expert people, can "fix"  
or

check a proxy configuration for a high profile network in 2 hours
(neither in 8 hours).
Such kind of miracle doesn't exist.


Then there's the financial issue.  People with solid knowledge are
usually quite busy;  there is no way it is worth anyone's time to  
drop

what they're doing and get involved in anything for 2 hours.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [OpenSIPS-Users] Paid Consultation Request

[Adrian Georgescu]

You will likely obtain more help from this mailing lists if you narrow  
down the scope of your questions and ask for help about the specific  
problems you encounter. For example locate your problem, describe it,  
provide a useful trace.


You will be surprised how effective this can be. Just a different  
angle of approach will yield more results. Or don't name that price ;-)


Adrian


On Feb 11, 2009, at 9:48 PM, Geoffrey Mina wrote:


There is no NAT
There is no Firewall (running local IPTables)
There are no 'users'
There are no 'phones'
There is a single upstream Carrier
There ARE a plurality of asterisk based IVR systems which are
processing the media.  All configured identically, all on the same
network.

I have been doing SIP for years, just not using OpenSIPS as my
Loadbalancer/Gateway of choice.
I have been writing networked applications for 5 times the length of
my SIP experience... an OpenSIPS N00B perhaps, but a network/SIP N00B
I think not.

I know enough to know that my configuration is NOT complex.  I am
shocked to hear that so many people on this list think this type of
OpenSIPS configuration is complicated to the point that an experienced
user couldn't take a look at a simple 150 line configuration file and
point out any glaring issues and identify some best practice
scenarios.

Hopefully there are some experienced users out there who are
interested in contributing something useful instead of this dribble
about how smart you are and how complex your OpenSIPS configuration
is.

-Geoff



On Wed, Feb 11, 2009 at 2:25 PM, John Rose   
wrote:
Often people and companies underestimate the complexities of a SIP  
proxy
installation. There are many variables particularly ones with  
carriers,
phones, NAT's, firewalls...  Many assume it is just configuration.  
A lot of

it is but unless you have worked it in depth you won't know.

Here is a free "blessing" - Seek and you shall find my dear  
OpenSIPS N00B.

Let the light guide you through the valley of SIP and networking
complexities :)

John


-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Geoffrey Mina
Sent: Wednesday, February 11, 2009 9:47 AM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Paid Consultation Request

Hello,
I am looking for anyone who would consider themselves an 'expert' in
the field of OpenSIPS.  My company is launching an OpenSIPS  
deployment

to front-end all the SIP traffic entering our network.  I would like
to have someone experienced look over my config to give it the
proverbial 'blessing'.

We run a very high profile network and I can't afford to have any
minor misconfigurations or problems cause issues down the road.

I would be willing to pay $150/hr (USD) via PayPal.  I am guessing I
will need 2(ish) hours, and IM/MSN/Skype chat would be the best.

If anyone is interested, please let me know.
Thanks,
Geoff

p.s. we are primarily using the dispatcher module, so any interested
party should know that one inside and out!

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Re: [OpenSIPS-Users] Business around OpenSIPS

[Adrian Georgescu]

Bogdan

I know that is a very hairy subject and we have precedents behind us  
that we can learn from.


An open Wiki page will likely yield, given enough time, on  
opensips.org the N-th directory listing of 'I am also here doing some  
xSER stuff'


I would rather make a page where relevant consultants /products are  
listed only based on recommandations. That is, only people and  
products recommended by other customers should be allowed. It would be  
great to find an innovative way to validate input based on some social  
networking criteria rather than a list me here button.



Adrian



On Feb 12, 2009, at 1:16 PM, Iñaki Baz Castillo wrote:


2009/2/12 Bogdan-Andrei Iancu :

2) a web page for listing business entities that may help you in
whatever OpenSIPS related issue. Keep in mind, this page is just a
listing (per sections) and not a place for publicity or service/ 
product

descriptions. The idea is simple: if you do business around OpenSIPS,
you may list yourself there for people to find you - nothing more.  
Page

is free to edit, so anybody can list himself:
http://www.opensips.org/pmwiki.php?n=Resources.Business


Can just "persons" appear in that list? (I mean withou a company).

--
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Re: [OpenSIPS-Users] CDRTool Issue

[Adrian Georgescu]

Hm, you must have provisioned something wrongly. Min duration as far  
as I remember has to do with the amount of seconds a call should have  
to be billed. Please reduce that value to 0 see if this solves it.


Adrian


On Feb 12, 2009, at 3:23 PM, Brian Chamberlain wrote:


Hi,

My apologies I investigate further I see this is for debugging  
reasons I assume:


Duration: 120 s App: audio Destination: 3531 
Customer: subscriber=1...@sw1.dub.asterisk.ieIncrement: 60 s Min  
duration: 300 s




My price is showing as 0 after renormalization. I have a rate for  
destination 3531, everything else seems fine. Any ideas?


Thanks,
Brian



On 12 Feb 2009, at 13:11, Brian Chamberlain wrote:


Hi All,

I have copied one of my radacct tables into radacct200807 to test  
my billing and rates. It is a table from January. When I  
renormalise for a day I don't get any errors for the customer that  
I have added, I have them pointing at a correct profile which in  
turn points at correct rates for the times of day. The records  
appear normalized and rated however the rate is zero.


If I look at the records the RATE field on each record seems to  
contain incorrect data. It contains:


 Duration: 120 s
App: audio
Destinati...

This record had a duration of 85s and I have set billing intervals  
to one minute.


How do I fix this?

Thanks,
Brian.

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Re: [OpenSIPS-Users] mediaproxy 2.3: reinvites' SDP not modified

[Adrian Georgescu]

Can you register two sip accounts on http://sip2sip.info

Then make a T.38 fax call and see if it works. If it does not, let me  
know the Call Id, there I have access to traces and we can find out  
the cause of your problem.


Adrian

On Feb 12, 2009, at 3:44 PM, Carlo Dimaggio wrote:



Il giorno 11/feb/09, alle ore 12:00, Ruud Klaver ha scritto:


Hi Carlo,

The problem definitely lies with OpenSIPS in your first example,  
the one that didn't work, since the media-dispatcher does not  
receive anything from the mediaproxy module in OpenSIPs when the re- 
INVITE is received. Could you please provide the relevant parts of  
your OpenSIPS config and its logging for both of these cases?


From the logs I gave you I cannot tell that much, except that in  
the first case your proxy seems to anonymize the SIP URI of the  
caller, but this shouldn't matter.


The mediaproxy module should just look for the m=image line, which  
is the same in both traces. Other than that it does no processing  
of the T38 stuff, it should be agnostic to it.


Hi Ruud,

Sure. I attach the relevant parts of my configuration and the log  
for both of these cases.


I don't understand why there is this behaviour with two devices  
(grandstream and patton) and the same opensips.cfg... In both of  
these cases, the re-invite is performed (as you can see from the  
ngrep log) but when I use the patton, it seems that mediaproxy  
doesn't recognise the t38 session and doesn't proxy the call.
I use engage_media_proxy() in my script that should automatically  
call use_media_proxy() on every request and reply that belongs to  
the dialog...


What kind of errors could be in my configuration?


Thank you,
Carlo Dimaggio

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Re: [OpenSIPS-Users] CDRTool Q

[Adrian Georgescu]

On Feb 12, 2009, at 5:53 PM, Brian Chamberlain wrote:


Hi Adrian,

I got my issue resolved with the rating not appearing. 2 Questions  
for you:




Two? :-)

Say a Provider operating and the normal rules are 0 for local 00 for  
international and the country code is 44


How should I write phone number dialled for a special rate number  
for example a fixed rate 1890 number. The customer dials 1890 .  
CDRTool logic will think this is america.


Well, it will not be able to find a destination ID for such a short  
number so it will not consider it as being an US destination.


Just put in an international rate even though it's special local  
dialcode? ie, pretend it's an American number.


Try entering u...@domain in the rate and dest table. The full URI  
taken from the CanonicalURI field will be matched, which is what you  
want for these service numbers.


Second question. When you renormalize through the web gui is there a  
maximum number of records you can do or some other criteria. It  
seems if I try to do above 7 days it will not work.


There is a section about re-normalization of CDRs in RATING.txt  
document. It may shed some light about this issue.




Thanks,
Brian





Brian Chamberlain
Dot Net Solutions Ltd.
68 Parkwest Enterprise Centre,
Parkwest,
Dublin 12,
Ireland.
DDI:
[+353]
1
6296521

FAX:
[+353]
1
6237029


mobile:
[+353]
86
3883003



web:
www.asterisk.ie

* Looking for the most advanced PBX available that can also save you  
a fortune in communication costs?  asterisk.ie *



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Re: [OpenSIPS-Users] Business around OpenSIPS

[Adrian Georgescu]

What I have noticed is that open wiki pages, polling and voting  
mechanisms tend to be easily polluted or misused. What can be faked  
with much more effort are:


1. Links to customer PR sites
2. Software that advertises that it uses or complements OpenSIPS
3. Third party blogs

Practically anything that indirectly point to a resource seem to be  
the way Google calculates relevance to a web resource. Maybe we can re- 
use this same logic google does?


This way we can still have the  open wiki page (easy entry) while  
adding a column with confirmations/recommendation from third parties  
(confirmation). Those with zero references are easily  spotted and can  
be demoted after a grace period of time.


Can this work in your opinion?


On Feb 12, 2009, at 7:33 PM, Bogdan-Andrei Iancu wrote:


Hi Adrian,

I agree that something like that will be good, but if you have an  
idea about how to implement it. I'm ready for suggestions :).


Regards,
Bogdan

Adrian Georgescu wrote:

Bogdan

I know that is a very hairy subject and we have precedents behind  
us that we can learn from.


An open Wiki page will likely yield, given enough time, on  
opensips.org the N-th directory listing of 'I am also here doing  
some xSER stuff'


I would rather make a page where relevant consultants /products are  
listed only based on recommandations. That is, only people and  
products recommended by other customers should be allowed. It would  
be great to find an innovative way to validate input based on some  
social networking criteria rather than a list me here button.



Adrian



On Feb 12, 2009, at 1:16 PM, Iñaki Baz Castillo wrote:

2009/2/12 Bogdan-Andrei Iancu mailto:bog...@voice-system.ro 
>>:

2) a web page for listing business entities that may help you in
whatever OpenSIPS related issue. Keep in mind, this page is just a
listing (per sections) and not a place for publicity or service/ 
product
descriptions. The idea is simple: if you do business around  
OpenSIPS,
you may list yourself there for people to find you - nothing  
more. Page

is free to edit, so anybody can list himself:
   http://www.opensips.org/pmwiki.php?n=Resources.Business


Can just "persons" appear in that list? (I mean withou a company).

--
Iñaki Baz Castillo
mailto:i...@aliax.net>>

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Re: [OpenSIPS-Users] CDRTool Q

[Adrian Georgescu]

>>> Say a Provider operating and the normal rules are 0 for local 00  
>>> for international and the country code is 44
>>>
>>> How should I write phone number dialled for a special rate number  
>>> for example a fixed rate 1890 number. The customer dials 1890 .  
>>> CDRTool logic will think this is america.
>>
>> Well, it will not be able to find a destination ID for such a short  
>> number so it will not consider it as being an US destination.
>>
>
> Sorry, I didn't explain very well, 1890 is the prefix, so it will be  
> 1890xx and that is just an example, there are also premium rate  
> numbers 1550xx for example which are charged at quite high rates..

The Canonical URI should always contain the full E164 number. From the  
point of view of the engine those numbers should be 441890X with  
country code in front.

The fact that your send those numbers without 44 is something you  
should rewrite in the request URI in the proxy when the Invite goes out.

> I couldn't see anything about any limitiation on the number of CDR's  
> that can be renormalized using the GUI.

It much depends how much time it takes. You do not want to re- 
normalize 1 million records from the web page. You should not use the  
web page for more than a few hundred records.

> It seems the only sure fire way that I can get them to renormalize  
> using the gui is to do it in 24h lumps. I can't use the php script  
> to do it, it will only do Jan/Feb and ignores radacct200812.
>
> Is there a correct place to submit feature requests?

Would an option for the normalize.php script to use a specific table  
solve this?

If you are in a hurry just edit the normalize.php script and set $CDRS- 
 >table=your_own_table before the normalize() function.

> One thing that would be really nice is if you are billing in /60 it  
> would be nice to have a period in, ie. if you are getting billed in / 
> 1 from your SIP providers your margin is incorrect if you are  
> billing /60.

I believe that what you describe is already possible or I do not  
undertand your question.

Adrian


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Re: [OpenSIPS-Users] CDRTool Q

[Adrian Georgescu]

On Feb 13, 2009, at 5:53 PM, Brian Chamberlain wrote:

> Hi Adrian,
>
> Thanks for the answers, spot on! One last question (pushing my luck)!
>
> If I want to give some customers different rates for a very small  
> number of destinations (like 3) how do I do this without having to  
> duplicate two entire sets of rates (on/off peak).

Set a rate plan with the 3 destinations than in the customer table set  
the fallback profile to your full table.

> Could I setup a new profile with point that profile at the special  
> rates with the default rates as fallback or is this a bad way of  
> doing it, ie., will rate the calls but generate a lot of errors?

Yes, every customer has a main and fallback profile so you can realize  
this setup out of the box.

Adrian


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Re: [OpenSIPS-Users] Mediaproxy

[Adrian Georgescu]

On Feb 16, 2009, at 4:14 AM, ram wrote:


Hi

Same thing i have complained to maintainer
its been 10days no mail from him

we need to just wait until we hear from them

Ram

On Sat, Feb 14, 2009 at 10:46 PM, NYam  wrote:
Dear Sir/Madam,

I have following issue and need to address to you.

I have operating system-debian R7, which installed kamailio 1.4.  
When i tried to install mediaproxy-2.3.2.tar.gz, it does not match  
because python should be equal or higher than 2.5. A


In the documentation is stated that you need Python 2.4.

Where do you see that it needs at least Python 2.5?

Adrian

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Re: [OpenSIPS-Users] Adtran VQM data into Opensips/CDRTool

[Adrian Georgescu]

If you saved that information CDRTool it will merely display it in the  
expanded CDR info, nothing more.


Adrian

On Feb 16, 2009, at 10:04 PM, Jeff Pyle wrote:


Hello,

We run a number of Adtran TA900-series gateways in our network.   
Recent software versions have the ability to do voice quality  
measurements, and output that data at the end of a call through a  
PUBLISH event.  Normally one would configure these devices to talk  
to an Adtran n-Command MSP data collector.


Because this PUBLISH happens after the call is disconnected, it  
seems an update would have to occur after the fact, perhaps similar  
to a Mediaproxy info update.  In the example below that Adtran  
provided to me, the CallID field is ‘unknown’.  I’m waiting on a  
response from Adtran engineering to see if that is always the case.   
If so, it’s going to be more difficult to match the call.


The following is from the perspective of the Adtran unit:

Tx: TCP src=10.19.209.11:5060 dst=10.100.13.250:5060
PUBLISH sip:collec...@10.100.13.250 SIP/2.0
From:  
;tag=4afcca8-0-13c4-3954f- 
f39d4b80-3954f

To: 
Call-ID: 4afcca8-0-13c4-3954f-e9ebf5f2-39...@10.19.209.11
CSeq: 1 PUBLISH
Via: SIP/2.0/TCP 10.19.209.11:5060;branch=z9hG4bK-3954f- 
dff3d63-6155485b

Event: vq-rtcpxr
Subscription-State: active
Content-Type: application/vq-rtcpxr
Content-Length: 1263

VQSessionReport
LocalMetrics:
Timestamps:START=2008-12-18T17:55:01Z STOP=2008-12-18T17:55:10Z
SessionDesc:PT=0 PC=1
CallID:unknown
LocalAddr:IP=10.19.209.55 PORT=3004 SSRC=22b4624f
RemoteAddr:IP=10.19.209.49 PORT=1 SSRC=22b4624f
JitterBuffer: JBRSYNC=5 JBP=463 JBPOO=0 JBPD=0 JBPE=37 JBPL=425  
JBRC=28 JBENVD=1 JBENVP=0 JBENVPMIN=0 JBENVPM=4 JBENVN=0 JBENVNMIN=0  
JBENVNM=0 JBLT=50.0 JBLTP=100.00 JBLUT=11 JBL=11 JBLPJ=2.0 JBET=10.0  
JBETP=100.00 JBE=451 JBEPJ=0.0 JBT=1 JBCDMIN=10 JBCDN=50 JBCDM=100  
JBDINC=0 JBDDEC=3 JBD=47 JBDI=35 JBDMIN=35 JBDM=50

PacketLoss:NLR=0.00 JDR=0.00 LR=0.00 JL=0 JD=0 JOD=0 JUD=0
BurstGapLoss:BLD=0.00 BD=0 GLD=0.00 GD=8640 BPD=0 BC=0 BE=0  
GPD=432 GC=1 GE=0
Delay:RTD=1 ESD=65 OWD=63 IAJ=451 RTDI=1 RTDM=1 ESDMIN=55  
ESDM=70 OWDI=58 OWDM=65 LD=0 LDMIN=0 LDM=2 PPDV=0.3 PDV=0 PDVM=2  
PDVMN=0.0 PDVMNI=0.4 PDVMNM=0.0 PDVMNAM=2.0

Signal:
QualityEst:RLQ=93 RCQ=92 MOSLQ=4.20 MOSCQ=4.18 BRLQ=92 GRLQ=92  
RN=93 RG107=92 MOSPQ=4.45 MOSN=4.20 QL=0

MetricsVersion:MT=ADTRAN MV=01.00
DeviceSerialNum:LBADTN0816AE392
CCMID:39
DSCP:46
LocalCaller:true
LocalURI:8249
LocalIfc:4
RemoteURI:8884238726
RemoteIfc:0
Degradation:DL=0.00 DDISC=0.00 DV=0.00 DR=0.00 DD=0.00 DSL=0.00  
DNL=0.00 DEL=1.42

Data:TI=35000 RI=100 BR=64000

Rx: TCP src=10.100.13.250:5060 dst=10.19.209.11:5060
SIP/2.0 200 Ok
Content-Length: 0
From:  
;tag=4afcca8-0-13c4-3954f- 
f39d4b80-3954f

Call-ID: 4afcca8-0-13c4-3954f-e9ebf5f2-39...@10.19.209.11
CSeq: 1 PUBLISH
Via: SIP/2.0/TCP 10.19.209.11:5060;branch=z9hG4bK-3954f- 
dff3d63-6155485b

To: ;tag=482617583
Allow: OPTIONS, PUBLISH
Allow-Events: vq-rtcpxr
SIP-ETag: 1229622954690.781
Expires: 0

First real question:  what’s the best way to get the  
“VQSessionReport” data out of this packet?  Some of the data is  
useful (particularly the MOSPQ value on the QualityEst line), much  
of it is not.  Unfortunately I don’t know anything about the normal  
uses for PUBLISH.


Assuming one can extract this useful information from this and match  
it to an existing call in radius and push the useful information  
into the RTPStatistics field, what would CDRTool do with it?



Thanks,
Jeff
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