opensipstack-devel

Re: [OpenSIPStack] SRTP

[Joegen E. B. <jo...@pl...>]

Yes, RTP and SIP are encrypted.

Leo Ann Boon wrote:
> Joegen,
>   
>> Unfortunately not.  The Encryption mentioned in the web site is a none 
>> standard simple encryption supported by Grand Stream and Inter Edge and 
>> its not SRTP.
>>   
>>     
> Can you elaborate a little more on this non-standard encryption used by 
> GS? does it encrypt the SIP and media packet?
>
> Leo
>
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

Re: [OpenSIPStack] SRTP

[Leo A. B. <le...@da...>]

Joegen,
> Unfortunately not.  The Encryption mentioned in the web site is a none 
> standard simple encryption supported by Grand Stream and Inter Edge and 
> its not SRTP.
>   
Can you elaborate a little more on this non-standard encryption used by 
GS? does it encrypt the SIP and media packet?

Leo

Re: [OpenSIPStack] SRTP

[Joegen E. B. <jo...@pl...>]

Leandro Caetano Gonçalves Lustosa wrote:
> Hi,
>
> Reading the features information on OSS website I noticed something
> regarded to "SIP/RTP Hash Encryption". Is it a SRTP and SIP over TLS
> implementation?
>
> Thanks,
>   
Hi Leandro,

Unfortunately not.  The Encryption mentioned in the web site is a none 
standard simple encryption supported by Grand Stream and Inter Edge and 
its not SRTP.

Joegen

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[tele <te...@pl...>]

Hi Joegen,

I'll do some test in the weekend.

:tele

On Tue, 2006-10-31 at 18:24 +0800, Joegen E. Baclor wrote:
> Hi Tele,
> 
> One more thing.  Please set m_DefaultSocket to NULL in the 
> constructor.   I've forgotten to initialize the thing!
> 
> SIPUDPSocketList::SIPUDPSocketList(
>   SIPTransportManager & mgr
> ) : m_TransportManager( mgr )
> {
>   m_DefaultSocket = NULL;  /// <<------------------------------
>   m_SocketList.DisallowDeleteObjects();
>   m_SelectList.DisallowDeleteObjects();
> }
> 
> Joegen
> 
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

[OpenSIPStack] SRTP

[<lca...@gm...>]

Hi,

Reading the features information on OSS website I noticed something
regarded to "SIP/RTP Hash Encryption". Is it a SRTP and SIP over TLS
implementation?

Thanks,
-- 
Leandro

Re: [OpenSIPStack] OpenSBC questions

[Joegen E. B. <jo...@pl...>]

Hi Tele,

There are only two places in the listener where a null socket may be 
returned.  Here's the other one.  If this doesn't fix it, i am going to 
need a complete stack trace if it's possible.

PUDPSocket * SIPUDPSocketList::GetSocket(
 const PIPSocket::Address & sendAddress
)
{
 if( m_SocketList.GetSize() == 0 )
   return NULL;

 PIPSocket::Address ifaceAddress = 0;
 if( !sendAddress.IsLoopback() )
 {

   ifaceAddress =
     SIPTransportManager::GetDefaultInterfaceAddress( sendAddress );
     if( !ifaceAddress.IsValid() || ifaceAddress.IsLoopback() )
     ifaceAddress = SIPTransport::GetDefaultInterfaceAddress();
   }else
 {
   ifaceAddress = PIPSocket::Address( "127.0.0.1" );;
 }


 PTRACE( 4, "Using Iface: " << ifaceAddress << " to send to Dest: " << 
sendAddress );
 PCaselessString key = ifaceAddress.AsString();
 PUDPSocket * sock = m_SocketList.GetAt( key );

/// use the default socket
 if( sock == NULL )
   sock = m_DefaultSocket;

/// still NULL then just return the first socket in the list
/// wasn't able to determine default socket???
 if( sock == NULL )
   sock = &m_SocketList[0];

 return sock;
}

Joegen

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[Joegen E. B. <jo...@pl...>]

Hi Tele,

One more thing.  Please set m_DefaultSocket to NULL in the 
constructor.   I've forgotten to initialize the thing!

SIPUDPSocketList::SIPUDPSocketList(
  SIPTransportManager & mgr
) : m_TransportManager( mgr )
{
  m_DefaultSocket = NULL;  /// <<------------------------------
  m_SocketList.DisallowDeleteObjects();
  m_SelectList.DisallowDeleteObjects();
}

Joegen

Re: [OpenSIPStack] OpenSBC questions

[tele <te...@pl...>]

I think that we will consider the commercial version from Solegy ;)
I'll contact Solegy if there are update on our project.

regards,


:tele


On Tue, 2006-10-31 at 00:35 +0800, Joegen E. Baclor wrote:
> >   
> Most definitely!   We have working commercial installations exactly as 
> you outlined.
> >
> >
> >
> >
> > -------------------------------------------------------------------------
> > Using Tomcat but need to do more? Need to support web services, security?
> > Get stuff done quickly with pre-integrated technology to make your job easier
> > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[tele <te...@pl...>]

Hi Joegen,

This doesn't fix problem it segfault in the same point.

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0x84e9c00 (LWP 100085)]
0x082b430e in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
1619      return port;
(gdb)

b2bua-log

2006/10/31 09:38:22.152 0x8425ea0       OSS DTL.......................|
Starting UDP Transport on 82.215.130.50:5061
2006/10/31 09:38:22.154 0x8425ea0       OSS DTL.......................|
*** TRANSPORT STARTED ***  Listener: 82.215.130.50:5061
2006/10/31 09:38:32.202 0x84f5880       OSS DBG.......................|
RCV: XOR=0 974 Bytes from RCVADDR: 82.215.130.50:RCVPORT: 5060:UDP
(INVITE sip:222@
82.215.130.50:5061;user=phone SIP/2.0)
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  RECEIVE:  REQ: INVITE (40362)
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  FROM:
sip:11...@sb...;user=phone
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  TO:
sip:22...@sb...;user=phone
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  XOR: FALSE
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  SRC: 82.215.130.50 PORT: 5060
2006/10/31 09:38:32.204 0x84f5880       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.204 0x84f5880       OSS
DBG......................................| Finding transaction for
INVITE sip:222@82.215.130.50:5061;user=phone
 SIP/2.0
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................| Setting Transaction ID to
4308362cf7db0e0c@82.215.128.141|z9hG4bK10f
f.d19711426464e0b03dac37c69454f869.0|INVITE
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................|  
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................|  *** CREATING TRANSACTION
(IST) ***
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................|  Message: INVITE
sip:222@82.215.130.50:5061;user=phone SIP/2.0
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................|  Call-Id:
4308362cf7db0e0c@82.215.128.141
2006/10/31 09:38:32.205 0x84f5880       OSS
DBG......................................|  
2006/10/31 09:38:32.205 0x84f5880       OSS
DTL......................................| IST(1162287512206)       ***
CREATED *** - IST|4308362cf7db0e0c@82.2
15.128.141|z9hG4bK10ff.d19711426464e0b03dac37c69454f869.0|INVITE
2006/10/31 09:38:32.206 0x8463100       OSS DTL:
[4308362cf7db0e0c@82.215.128.141]...| IST(1162287512206)
Event(SIPMessage) - INVITE sip:222@82.215.1
30.50:5061;user=phone SIP/2.0
2006/10/31 09:38:32.206 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...| TRANSACTION: (IST) INVITE
sip:222@82.215.130.50:5061;user=phone SIP/
2.0 State: 0
2006/10/31 09:38:32.206 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.206 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  *** IST STATE IDLE ***
2006/10/31 09:38:32.206 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  INVITE
sip:222@82.215.130.50:5061;user=phone SIP/2.0
2006/10/31 09:38:32.206 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  TRANSMIT: RESPONSE: SIP/2.0 100
Trying
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  TO:
sip:22...@sb...;user=phone
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  REQ: INVITE (40362)
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  XOR: FALSE
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  DST: 82.215.130.50 PORT: 5060
2006/10/31 09:38:32.208 0x8463100       OSS DBG:
[4308362cf7db0e0c@82.215.128.141]...|  
2006/10/31 09:38:32.210 0x8463100       OSS DTL:
[4308362cf7db0e0c@82.215.128.141]...| IST(1162287512206)
StateIdle->StateProceeding
2006/10/31 09:38:32.211 0x84f5980       OSS
DBG......................................| SEND: XOR=0 399 Bytes to
82.215.130.50:5060:UDP (SIP/2.0 100 Trying)


thank you

:tele



On Tue, 2006-10-31 at 10:27 +0800, Joegen E. Baclor wrote:
> Hi Tele,
> 
> Try replacing GetListenerAddress() method in SIPUDPSocketList with the 
> code below.  Let me know if that fixes the problem.
> 
> BOOL SIPUDPSocketList::GetListenerAddress(
>   const PIPSocket::Address & sendAddress,
>   PIPSocket::Address & ifaceAddress,
>   WORD & listenerPort
> )
> {
>   if( m_SocketList.GetSize() == 0 )
>     return FALSE;
> 
>   if( !sendAddress.IsLoopback() )
>   {
> 
>     ifaceAddress =
>       SIPTransportManager::GetDefaultInterfaceAddress( sendAddress );
>    
>     if( !ifaceAddress.IsValid() || ifaceAddress.IsLoopback() )
>       return FALSE;
>    
>   }else
>   {
>     ifaceAddress = PIPSocket::Address( "127.0.0.1" );;
>   }
> 
>   PCaselessString key = ifaceAddress.AsString();
>   PUDPSocket * sock = m_SocketList.GetAt( key );
> 
>   if( sock == NULL )
>   {
>     ifaceAddress =
>       SIPTransportManager::GetDefaultInterfaceAddress();
>    
>     if( sock == NULL )
>       return  FALSE;
> 
>     sock = m_SocketList.GetAt( key );
> 
>     if( sock == NULL )
>       sock = &m_SocketList[0];
> 
>     if( !ifaceAddress.IsValid() && !sock->GetHostAddress( ifaceAddress ) )
>       return FALSE;
>    
>   }
>  
>   listenerPort = sock->GetPort();
> 
>   return TRUE;
> }
> 
> 
> tele wrote:
> > OpenSIPStack use his pwlib tree
> >
> > set -e; gmake -C src/pwlib opt; gmake -C src opt;
> > gmake[1]: Entering directory `/usr/local/src/opensipstack/src/pwlib'
> > set -e; gmake -C src/ptlib/unix opt; gmake -C plugins opt;
> > gmake[2]: Entering directory
> > `/usr/local/src/opensipstack/src/pwlib/src/ptlib/unix'
> >
> > now i've uninstall the separately pwlib library and recompiling the
> > OpenSIPstack.
> >
> > anyway the problem with segfault is on:
> >
> > line 1619 of 
> > /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
> >
> > attached you can find the b2bua-log.
> >
> >
> > On Mon, 2006-10-30 at 23:53 +0800, Joegen E. Baclor wrote:
> >   
> >> Hi Tele,
> >>
> >> Thanks for the bug report.  First, let's eliminate the possibility that 
> >> this is a version conflict between different pwlib installation in your 
> >> box.  You said you compiled pwlib separately?   If so please delete all 
> >> other instances of pwlib in your file system and let opensipstack use 
> >> the pwlib that goes with the source tree.  Do a rerun and see if it 
> >> still seg faults.  If it does, I am going to need level 5 logs of 
> >> b2bua-yy-mm-dd.log.  You can set the log level using the http admin 
> >> listening at port 9999.   Look for the OpenSBC General Parameters 
> >> link.   If you can find out where the exact line number and which file 
> >> causes the seg fault, so much the better.  Hope to hear more from you.
> >>
> >> Joegen
> >>
> >> tele wrote:
> >>     
> >>> Hello,
> >>>
> >>> I trying to use OpenSBC in a FreeBSD system.
> >>>
> >>> I've compiled opensipstack and opensbc successful.
> >>> I run opensbc and when the first INVITE come it's segfault.
> >>>
> >>> i've compiled opensbc with gmake debugnoshared and this is the gdb
> >>> backtrace:
> >>>
> >>> # gcc -v
> >>> Using built-in specs.
> >>> Configured with: FreeBSD/i386 system compiler
> >>> Thread model: posix
> >>> gcc version 3.4.4 [FreeBSD] 20050518
> >>>
> >>>
> >>>
> >>> (gdb) r -x
> >>> Starting program: /usr/local/bin/opensbc -x
> >>> warning: Unable to get location for thread creation breakpoint: generic
> >>> error
> >>> [New LWP 100143]
> >>> [New Thread 0x8438000 (LWP 100143)]
> >>>
> >>> Message from syslogd@sbc01 at Mon Oct 30 13:25:54 2006 ...
> >>> sbc01 OpenSIPStack Application[82385]: Starting service process
> >>> "OpenSIPStack Application" v1.1.3
> >>>
> >>>
> >>> [New Thread 0x84e8a00 (LWP 100142)]
> >>>
> >>> Program received signal SIGSEGV, Segmentation fault.
> >>> [Switching to Thread 0x84e8a00 (LWP 100142)]
> >>> 0x082b429a in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
> >>> 1619      return port;
> >>> (gdb) 
> >>>
> >>>
> >>> The segfault is caused by the pwlib 
> >>> /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
> >>>
> >>> but i've compiled separately pwlib and export the enviroment vars, how
> >>> say the ReadMe.txt of opensipstack.
> >>>
> >>> regards,
> >>>
> >>>
> >>> :tele
> >>>
> >>>
> >>>
> >>> -------------------------------------------------------------------------
> >>> Using Tomcat but need to do more? Need to support web services, security?
> >>> Get stuff done quickly with pre-integrated technology to make your job easier
> >>> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> >>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> >>> _______________________________________________
> >>> opensipstack-devel mailing list
> >>> ope...@li...
> >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >>>
> >>>   
> >>>       
> >> -------------------------------------------------------------------------
> >> Using Tomcat but need to do more? Need to support web services, security?
> >> Get stuff done quickly with pre-integrated technology to make your job easier
> >> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> >> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> >> _______________________________________________
> >> opensipstack-devel mailing list
> >> ope...@li...
> >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >>     
> >> ------------------------------------------------------------------------
> >>
> >> 2006/10/30 16:58:27.728	0x8425e00	OSS DTL.......................| Starting UDP Transport on 82.215.130.50:5061
> >> 2006/10/30 16:58:27.730	0x8425e00	OSS DTL.......................| *** TRANSPORT STARTED ***  Listener: 82.215.130.50:5061
> >> 2006/10/30 16:58:41.714	0x84f5880	OSS DBG.......................| RCV: XOR=0 974 Bytes from RCVADDR: 82.215.130.50:RCVPORT: 5060:UDP (INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0)
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	RECEIVE:  REQ: INVITE (49005)
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	FROM: sip:11...@sb...;user=phone
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TO: sip:22...@sb...;user=phone
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	XOR: FALSE
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	SRC: 82.215.130.50 PORT: 5060
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| Finding transaction for INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| Setting Transaction ID to 983d90ae1e37ba66@82.215.128.141|z9hG4bKbc98.7abb12a56fef89f4586a70cc43ea1ff2.0|INVITE
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................|  
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	*** CREATING TRANSACTION (IST) ***
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	Message: INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	Call-Id: 983d90ae1e37ba66@82.215.128.141
> >> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................|  
> >> 2006/10/30 16:58:41.718	0x84f5880	OSS DTL......................................| IST(1162227521719)	*** CREATED *** - IST|983d90ae1e37ba66@82.215.128.141|z9hG4bKbc98.7abb12a56fef89f4586a70cc43ea1ff2.0|INVITE
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| IST(1162227521719)	Event(SIPMessage) - INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| TRANSACTION: (IST) INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0 State: 0
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	*** IST STATE IDLE ***
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
> >> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TRANSMIT: RESPONSE: SIP/2.0 100 Trying
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TO: sip:22...@sb...;user=phone
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	REQ: INVITE (49005)
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	XOR: FALSE
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	DST: 82.215.130.50 PORT: 5060
> >> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
> >> 2006/10/30 16:58:41.722	0x8463100	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| IST(1162227521719)	StateIdle->StateProceeding
> >> 2006/10/30 16:58:41.724	0x84f5980	OSS DBG......................................| SEND: XOR=0 399 Bytes to 82.215.130.50:5060:UDP (SIP/2.0 100 Trying)
> >> 2006/10/30 16:58:41.726	0x84dc700	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| Event: ---> Inbound - INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
> >> 2006/10/30 16:58:41.726	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141]...| Session CREATED
> >> 2006/10/30 16:58:41.726	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141]...| *** CREATED *** Call Session
> >> 2006/10/30 16:58:41.727	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141-connection]...| Multidirectional Session CREATED
> >> 2006/10/30 16:58:41.727	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141-connection]...| B2BUAConnection Created 0x0x8516000 (Active Count: 1)
> >>     
> >> ------------------------------------------------------------------------
> >>
> >> -------------------------------------------------------------------------
> >> Using Tomcat but need to do more? Need to support web services, security?
> >> Get stuff done quickly with pre-integrated technology to make your job easier
> >> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> >> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> opensipstack-devel mailing list
> >> ope...@li...
> >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >>     
> 
> 
> 
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[Joegen E. B. <jo...@pl...>]

Hi Tele,

Try replacing GetListenerAddress() method in SIPUDPSocketList with the 
code below.  Let me know if that fixes the problem.

BOOL SIPUDPSocketList::GetListenerAddress(
  const PIPSocket::Address & sendAddress,
  PIPSocket::Address & ifaceAddress,
  WORD & listenerPort
)
{
  if( m_SocketList.GetSize() == 0 )
    return FALSE;

  if( !sendAddress.IsLoopback() )
  {

    ifaceAddress =
      SIPTransportManager::GetDefaultInterfaceAddress( sendAddress );
   
    if( !ifaceAddress.IsValid() || ifaceAddress.IsLoopback() )
      return FALSE;
   
  }else
  {
    ifaceAddress = PIPSocket::Address( "127.0.0.1" );;
  }

  PCaselessString key = ifaceAddress.AsString();
  PUDPSocket * sock = m_SocketList.GetAt( key );

  if( sock == NULL )
  {
    ifaceAddress =
      SIPTransportManager::GetDefaultInterfaceAddress();
   
    if( sock == NULL )
      return  FALSE;

    sock = m_SocketList.GetAt( key );

    if( sock == NULL )
      sock = &m_SocketList[0];

    if( !ifaceAddress.IsValid() && !sock->GetHostAddress( ifaceAddress ) )
      return FALSE;
   
  }
 
  listenerPort = sock->GetPort();

  return TRUE;
}


tele wrote:
> OpenSIPStack use his pwlib tree
>
> set -e; gmake -C src/pwlib opt; gmake -C src opt;
> gmake[1]: Entering directory `/usr/local/src/opensipstack/src/pwlib'
> set -e; gmake -C src/ptlib/unix opt; gmake -C plugins opt;
> gmake[2]: Entering directory
> `/usr/local/src/opensipstack/src/pwlib/src/ptlib/unix'
>
> now i've uninstall the separately pwlib library and recompiling the
> OpenSIPstack.
>
> anyway the problem with segfault is on:
>
> line 1619 of 
> /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
>
> attached you can find the b2bua-log.
>
>
> On Mon, 2006-10-30 at 23:53 +0800, Joegen E. Baclor wrote:
>   
>> Hi Tele,
>>
>> Thanks for the bug report.  First, let's eliminate the possibility that 
>> this is a version conflict between different pwlib installation in your 
>> box.  You said you compiled pwlib separately?   If so please delete all 
>> other instances of pwlib in your file system and let opensipstack use 
>> the pwlib that goes with the source tree.  Do a rerun and see if it 
>> still seg faults.  If it does, I am going to need level 5 logs of 
>> b2bua-yy-mm-dd.log.  You can set the log level using the http admin 
>> listening at port 9999.   Look for the OpenSBC General Parameters 
>> link.   If you can find out where the exact line number and which file 
>> causes the seg fault, so much the better.  Hope to hear more from you.
>>
>> Joegen
>>
>> tele wrote:
>>     
>>> Hello,
>>>
>>> I trying to use OpenSBC in a FreeBSD system.
>>>
>>> I've compiled opensipstack and opensbc successful.
>>> I run opensbc and when the first INVITE come it's segfault.
>>>
>>> i've compiled opensbc with gmake debugnoshared and this is the gdb
>>> backtrace:
>>>
>>> # gcc -v
>>> Using built-in specs.
>>> Configured with: FreeBSD/i386 system compiler
>>> Thread model: posix
>>> gcc version 3.4.4 [FreeBSD] 20050518
>>>
>>>
>>>
>>> (gdb) r -x
>>> Starting program: /usr/local/bin/opensbc -x
>>> warning: Unable to get location for thread creation breakpoint: generic
>>> error
>>> [New LWP 100143]
>>> [New Thread 0x8438000 (LWP 100143)]
>>>
>>> Message from syslogd@sbc01 at Mon Oct 30 13:25:54 2006 ...
>>> sbc01 OpenSIPStack Application[82385]: Starting service process
>>> "OpenSIPStack Application" v1.1.3
>>>
>>>
>>> [New Thread 0x84e8a00 (LWP 100142)]
>>>
>>> Program received signal SIGSEGV, Segmentation fault.
>>> [Switching to Thread 0x84e8a00 (LWP 100142)]
>>> 0x082b429a in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
>>> 1619      return port;
>>> (gdb) 
>>>
>>>
>>> The segfault is caused by the pwlib 
>>> /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
>>>
>>> but i've compiled separately pwlib and export the enviroment vars, how
>>> say the ReadMe.txt of opensipstack.
>>>
>>> regards,
>>>
>>>
>>> :tele
>>>
>>>
>>>
>>> -------------------------------------------------------------------------
>>> Using Tomcat but need to do more? Need to support web services, security?
>>> Get stuff done quickly with pre-integrated technology to make your job easier
>>> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
>>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>>   
>>>       
>> -------------------------------------------------------------------------
>> Using Tomcat but need to do more? Need to support web services, security?
>> Get stuff done quickly with pre-integrated technology to make your job easier
>> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>     
>> ------------------------------------------------------------------------
>>
>> 2006/10/30 16:58:27.728	0x8425e00	OSS DTL.......................| Starting UDP Transport on 82.215.130.50:5061
>> 2006/10/30 16:58:27.730	0x8425e00	OSS DTL.......................| *** TRANSPORT STARTED ***  Listener: 82.215.130.50:5061
>> 2006/10/30 16:58:41.714	0x84f5880	OSS DBG.......................| RCV: XOR=0 974 Bytes from RCVADDR: 82.215.130.50:RCVPORT: 5060:UDP (INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0)
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	RECEIVE:  REQ: INVITE (49005)
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	FROM: sip:11...@sb...;user=phone
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TO: sip:22...@sb...;user=phone
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	XOR: FALSE
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	SRC: 82.215.130.50 PORT: 5060
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| Finding transaction for INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| Setting Transaction ID to 983d90ae1e37ba66@82.215.128.141|z9hG4bKbc98.7abb12a56fef89f4586a70cc43ea1ff2.0|INVITE
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................|  
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	*** CREATING TRANSACTION (IST) ***
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	Message: INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................| 	Call-Id: 983d90ae1e37ba66@82.215.128.141
>> 2006/10/30 16:58:41.717	0x84f5880	OSS DBG......................................|  
>> 2006/10/30 16:58:41.718	0x84f5880	OSS DTL......................................| IST(1162227521719)	*** CREATED *** - IST|983d90ae1e37ba66@82.215.128.141|z9hG4bKbc98.7abb12a56fef89f4586a70cc43ea1ff2.0|INVITE
>> 2006/10/30 16:58:41.719	0x8463100	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| IST(1162227521719)	Event(SIPMessage) - INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
>> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| TRANSACTION: (IST) INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0 State: 0
>> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	*** IST STATE IDLE ***
>> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
>> 2006/10/30 16:58:41.719	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TRANSMIT: RESPONSE: SIP/2.0 100 Trying
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	TO: sip:22...@sb...;user=phone
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	REQ: INVITE (49005)
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	XOR: FALSE
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...| 	DST: 82.215.130.50 PORT: 5060
>> 2006/10/30 16:58:41.721	0x8463100	OSS DBG: [983d90ae1e37ba66@82.215.128.141]...|  
>> 2006/10/30 16:58:41.722	0x8463100	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| IST(1162227521719)	StateIdle->StateProceeding
>> 2006/10/30 16:58:41.724	0x84f5980	OSS DBG......................................| SEND: XOR=0 399 Bytes to 82.215.130.50:5060:UDP (SIP/2.0 100 Trying)
>> 2006/10/30 16:58:41.726	0x84dc700	OSS DTL: [983d90ae1e37ba66@82.215.128.141]...| Event: ---> Inbound - INVITE sip:222@82.215.130.50:5061;user=phone SIP/2.0
>> 2006/10/30 16:58:41.726	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141]...| Session CREATED
>> 2006/10/30 16:58:41.726	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141]...| *** CREATED *** Call Session
>> 2006/10/30 16:58:41.727	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141-connection]...| Multidirectional Session CREATED
>> 2006/10/30 16:58:41.727	0x84dc700	OSS INF: [983d90ae1e37ba66@82.215.128.141-connection]...| B2BUAConnection Created 0x0x8516000 (Active Count: 1)
>>     
>> ------------------------------------------------------------------------
>>
>> -------------------------------------------------------------------------
>> Using Tomcat but need to do more? Need to support web services, security?
>> Get stuff done quickly with pre-integrated technology to make your job easier
>> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>     

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[tele <te...@pl...>]

OpenSIPStack use his pwlib tree

set -e; gmake -C src/pwlib opt; gmake -C src opt;
gmake[1]: Entering directory `/usr/local/src/opensipstack/src/pwlib'
set -e; gmake -C src/ptlib/unix opt; gmake -C plugins opt;
gmake[2]: Entering directory
`/usr/local/src/opensipstack/src/pwlib/src/ptlib/unix'

now i've uninstall the separately pwlib library and recompiling the
OpenSIPstack.

anyway the problem with segfault is on:

line 1619 of 
/usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx

attached you can find the b2bua-log.


On Mon, 2006-10-30 at 23:53 +0800, Joegen E. Baclor wrote:
> Hi Tele,
> 
> Thanks for the bug report.  First, let's eliminate the possibility that 
> this is a version conflict between different pwlib installation in your 
> box.  You said you compiled pwlib separately?   If so please delete all 
> other instances of pwlib in your file system and let opensipstack use 
> the pwlib that goes with the source tree.  Do a rerun and see if it 
> still seg faults.  If it does, I am going to need level 5 logs of 
> b2bua-yy-mm-dd.log.  You can set the log level using the http admin 
> listening at port 9999.   Look for the OpenSBC General Parameters 
> link.   If you can find out where the exact line number and which file 
> causes the seg fault, so much the better.  Hope to hear more from you.
> 
> Joegen
> 
> tele wrote:
> > Hello,
> >
> > I trying to use OpenSBC in a FreeBSD system.
> >
> > I've compiled opensipstack and opensbc successful.
> > I run opensbc and when the first INVITE come it's segfault.
> >
> > i've compiled opensbc with gmake debugnoshared and this is the gdb
> > backtrace:
> >
> > # gcc -v
> > Using built-in specs.
> > Configured with: FreeBSD/i386 system compiler
> > Thread model: posix
> > gcc version 3.4.4 [FreeBSD] 20050518
> >
> >
> >
> > (gdb) r -x
> > Starting program: /usr/local/bin/opensbc -x
> > warning: Unable to get location for thread creation breakpoint: generic
> > error
> > [New LWP 100143]
> > [New Thread 0x8438000 (LWP 100143)]
> >
> > Message from syslogd@sbc01 at Mon Oct 30 13:25:54 2006 ...
> > sbc01 OpenSIPStack Application[82385]: Starting service process
> > "OpenSIPStack Application" v1.1.3
> >
> >
> > [New Thread 0x84e8a00 (LWP 100142)]
> >
> > Program received signal SIGSEGV, Segmentation fault.
> > [Switching to Thread 0x84e8a00 (LWP 100142)]
> > 0x082b429a in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
> > 1619      return port;
> > (gdb) 
> >
> >
> > The segfault is caused by the pwlib 
> > /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
> >
> > but i've compiled separately pwlib and export the enviroment vars, how
> > say the ReadMe.txt of opensipstack.
> >
> > regards,
> >
> >
> > :tele
> >
> >
> >
> > -------------------------------------------------------------------------
> > Using Tomcat but need to do more? Need to support web services, security?
> > Get stuff done quickly with pre-integrated technology to make your job easier
> > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
> >   
> 
> 
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Re: [OpenSIPStack] OpenSBC questions

[Joegen E. B. <jo...@pl...>]

Hi Tele,

inline....

tele wrote:
> Hello,
>
> Before starts i want to say thank you for the useful OpenSIPStack
> library.
> I've some question about the use of the OpenSBC "Reference" Application.
>
> I know it's only a "Reference" implementation but what are the
> functionality of the OpenSBC application?
>
> with the default installation it act as a normal SIP proxy and do relay
> based on regexp on the request uri.
>
>   

OpenSBC is a hybrid SIP Proxy and B2BUA.  The default behavior would be 
if a call  reaches  OpenSBC, i may branch out using separate logic 
depending on whether the call  resolves as a local domain (in which case 
OpenSBC will assume B2BUA logic) or resolves as a remote domain in which 
case OpenSBC would act as relay.   There are separate route provision 
for both modes in the HTTP admin.


> What about the RTBEClient missing ?
>
> checking for RTBESession.h... no
> RTBEClient support DISABLED...
>
> there is no RTBESession.h in the opensbc dir, is it required for the
> b2bua functionatility?
>   
RTBE stands for Real Time Billing Engine.  This is a proprietary client 
for the Solegy System's Billing backend.  It will also be out as an 
opensource component for OpenSBC contributed through 
http://www.opensourcesip.org.   RTBE client does not affect any 
functionality of OpenSBC 


> If act as a hybrid B2BUA/SIP Proxy it's possibile to achieve the
> Topology Hiding at the edge of network.
>
> What i want to know if it's possibile to achieve something like this
> with OpenSBC.
>
> A combination of OpenSBC/SER.
>
>             z
>          OpenSBC 
> EXT <------SER --------> INTERNAL
>           x   y        i
>
> In OpenSBC have a static routing based on src ip matching, something
> like that:
>
> if src_ip == y
>    relay_to x
> if src_ip == x
>    relay_to y
>
> so we have OpenSBC(b2bua) that connect the call-leg of call
> internal/external and we achieve the topology hiding of the internal
> network.
>
> It's not clear what i'm talking about ;)
>   
Most definitely!   We have working commercial installations exactly as 
you outlined.

> Is there a Roadmap for the OpenSBC application?
>   

Unfortunately, opensipstack is currently more of a one man team.  Most 
of my time is currently dedicated to coding.   So the answer is none 
yet.  Perhaps i'll be lucky and find that impossible moment where i drop 
the developer hat and put on the literary one  ;-)
> regards,
>
> :tele
>
>
>
>
>
>
>
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

[OpenSIPStack] OpenSBC questions

[tele <te...@pl...>]

Hello,

Before starts i want to say thank you for the useful OpenSIPStack
library.
I've some question about the use of the OpenSBC "Reference" Application.

I know it's only a "Reference" implementation but what are the
functionality of the OpenSBC application?

with the default installation it act as a normal SIP proxy and do relay
based on regexp on the request uri.

What about the RTBEClient missing ?

checking for RTBESession.h... no
RTBEClient support DISABLED...

there is no RTBESession.h in the opensbc dir, is it required for the
b2bua functionatility?

If act as a hybrid B2BUA/SIP Proxy it's possibile to achieve the
Topology Hiding at the edge of network.

What i want to know if it's possibile to achieve something like this
with OpenSBC.

A combination of OpenSBC/SER.

            z
         OpenSBC 
EXT <------SER --------> INTERNAL
          x   y        i

In OpenSBC have a static routing based on src ip matching, something
like that:

if src_ip == y
   relay_to x
if src_ip == x
   relay_to y

so we have OpenSBC(b2bua) that connect the call-leg of call
internal/external and we achieve the topology hiding of the internal
network.

It's not clear what i'm talking about ;)

Is there a Roadmap for the OpenSBC application?

regards,

:tele

Re: [OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[Joegen E. B. <jo...@pl...>]

Hi Tele,

Thanks for the bug report.  First, let's eliminate the possibility that 
this is a version conflict between different pwlib installation in your 
box.  You said you compiled pwlib separately?   If so please delete all 
other instances of pwlib in your file system and let opensipstack use 
the pwlib that goes with the source tree.  Do a rerun and see if it 
still seg faults.  If it does, I am going to need level 5 logs of 
b2bua-yy-mm-dd.log.  You can set the log level using the http admin 
listening at port 9999.   Look for the OpenSBC General Parameters 
link.   If you can find out where the exact line number and which file 
causes the seg fault, so much the better.  Hope to hear more from you.

Joegen

tele wrote:
> Hello,
>
> I trying to use OpenSBC in a FreeBSD system.
>
> I've compiled opensipstack and opensbc successful.
> I run opensbc and when the first INVITE come it's segfault.
>
> i've compiled opensbc with gmake debugnoshared and this is the gdb
> backtrace:
>
> # gcc -v
> Using built-in specs.
> Configured with: FreeBSD/i386 system compiler
> Thread model: posix
> gcc version 3.4.4 [FreeBSD] 20050518
>
>
>
> (gdb) r -x
> Starting program: /usr/local/bin/opensbc -x
> warning: Unable to get location for thread creation breakpoint: generic
> error
> [New LWP 100143]
> [New Thread 0x8438000 (LWP 100143)]
>
> Message from syslogd@sbc01 at Mon Oct 30 13:25:54 2006 ...
> sbc01 OpenSIPStack Application[82385]: Starting service process
> "OpenSIPStack Application" v1.1.3
>
>
> [New Thread 0x84e8a00 (LWP 100142)]
>
> Program received signal SIGSEGV, Segmentation fault.
> [Switching to Thread 0x84e8a00 (LWP 100142)]
> 0x082b429a in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
> 1619      return port;
> (gdb) 
>
>
> The segfault is caused by the pwlib 
> /usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx
>
> but i've compiled separately pwlib and export the enviroment vars, how
> say the ReadMe.txt of opensipstack.
>
> regards,
>
>
> :tele
>
>
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

[OpenSIPStack] OpenSBC segfault on FreeBSD 6.1

[tele <te...@pl...>]

Hello,

I trying to use OpenSBC in a FreeBSD system.

I've compiled opensipstack and opensbc successful.
I run opensbc and when the first INVITE come it's segfault.

i've compiled opensbc with gmake debugnoshared and this is the gdb
backtrace:

# gcc -v
Using built-in specs.
Configured with: FreeBSD/i386 system compiler
Thread model: posix
gcc version 3.4.4 [FreeBSD] 20050518



(gdb) r -x
Starting program: /usr/local/bin/opensbc -x
warning: Unable to get location for thread creation breakpoint: generic
error
[New LWP 100143]
[New Thread 0x8438000 (LWP 100143)]

Message from syslogd@sbc01 at Mon Oct 30 13:25:54 2006 ...
sbc01 OpenSIPStack Application[82385]: Starting service process
"OpenSIPStack Application" v1.1.3


[New Thread 0x84e8a00 (LWP 100142)]

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0x84e8a00 (LWP 100142)]
0x082b429a in PSocket::GetPort (this=0x0) at ../common/sockets.cxx:1619
1619      return port;
(gdb) 


The segfault is caused by the pwlib 
/usr/local/src/opensipstack/src/pwlib/src/ptlib/common/sockets.cxx

but i've compiled separately pwlib and export the enviroment vars, how
say the ReadMe.txt of opensipstack.

regards,


:tele

[OpenSIPStack] OSSPhone is now in CVS

[Joegen E. B. <jo...@pl...>]

All,

For those who are awaiting OSSPhone, it can now be downloaded from CVS 
together with the ATLSIP source tree.  It is only available as a VC++ 
8.0 project.   The download links will be updated accordingly in the  
next few days together with binary versions of OSSPhone, ATLSIP and OpenSBC.

Joegen

Re: [OpenSIPStack] ATLSIP - How to register multiple acccounts

[Borislav P. <bor...@ab...>]

Thanks Joegen. I will try it right away :)

-----------------------------------------------------------------
Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=E6=E0=E2=
=E0 =ED=E0 www.survivor.btv.bg=20

Re: [OpenSIPStack] ATLSIP - How to register multiple acccounts

[Joegen E. B. <jo...@pl...>]

Hi Borislav,

Until it finds its way in CVS, you can download a copy of OSSPhone 
here:  http://www.opensipstack.org/public/downloads/OSSPhone.zip.   This 
would only compile using visual studio 8.  You can use the FREE express 
edition to compile.   I hope it would help you get that Delphi Softphone 
you talked about out of the hole ;-)

Joegen


bor...@ma... wrote:
> Thanks, I understood how to register multiple users.
> I am a delphi developer and i would like to participate in the project 
> by translating the ATLSIP example to Delphi (the one that will come 
> out next week)
> > Hi,
> >
> > Currently ATLSIP only support single contact for
> > registrations.
> > Although it is relatively easy to modify this behavior
> > and be able to
> > send multiple contacts in a single REGISTER message.
> >
> > You need to overload the following method in opensipstack
> > library to
> > accept multiple users
> >
> > BOOL OpalOSSEndPoint::SendRegister(
> > const PString & userName,
> > const PString & password,
> > const PString & uri,
> > const PString & _domain,
> > BOOL synchronous
> > )
> >
> > I am currently working on a complete Softphone
> > application that
> > demonstrates in full how to use ATLSIP. It should be out
> > by next week.
> >
> > Joegen
> >
> > bor...@ma... wrote:
> > > I am trying to use ATLSIP in a program which must be
> > able to register
> > > a number of accounts and be able to handle incomming
> > calls for all
> > > those accounts.
> > > Could someone show me how can i do this?
> > > 10xs in advance
> > > -----------------------------
> > > Участвайте в онлайн анкетата и спечелете >>
> > > >> спортни обувки, екипи за тенис, суичери, спортни
> > сакове и още много
> > > награди
> > > предоставени от италианската марка Lotto и веригата
> > магазини за
> > > спортни стоки "Спорт Депо" ЕООД!
> > > Регистрирайте се още СЕГА: www.sportdepot.bg
> > >
> > 
> <http://mail.bg/ads/adclick.php?bannerid=4870&zoneid=13&source=&dest=http%3A%2F%2Fwww.sportdepot.bg%2F&ismap=> 
>
> > >
> > >
> > 
> ------------------------------------------------------------------------
> > >
> > >
> > 
> -------------------------------------------------------------------------
> > > Take Surveys. Earn Cash. Influence the Future of IT
> > > Join SourceForge.net's Techsay panel and you'll get the
> > chance to share your
> > > opinions on IT & business topics through brief surveys
> > -- and earn cash
> > >
> > 
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> > >
> > 
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > opensipstack-devel mailing list
> > > ope...@li...
> > >
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >
> >
> >
> > 
> -------------------------------------------------------------------------
> > Take Surveys. Earn Cash. Influence the Future of IT
> > Join SourceForge.net's Techsay panel and you'll get the
> > chance to share your
> > opinions on IT & business topics through brief surveys --
> > and earn cash
> > 
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
>
>
>
> -----------------------------
> Спортни залагания!
> bg.sportingbet.com 
> <http://mail.bg/ads/adclick.php?bannerid=4763&zoneid=13&source=&dest=http%3A%2F%2Fbg.sportingbet.com%2Fcontent%2Fhome.asp%3Faffiliate%3DMailbg&ismap=> 
>

Re: [OpenSIPStack] OpenSipStack

[Joegen E. B. <jo...@pl...>]

Hi Borislav,

Got the files, thanks.  I will let you know as soon as I've found the 
time to test it.

Joegen

Borislav Pechenyashki wrote:
> Hi,
> I am sending you a test project to reproduce the exception.
> My system is Windows XP Media Center and I use MSVS 2003 and Opensipstack 1.1.2 (I tried with version 1.1.3 and the latest one on CVS and the result is the same). I receive the exception either on program exit or on destruction of the MessengerManager. There are two files exception.txt and exception2.txt in which i have saved 2 of the exception that happended to me. But sometimes the excepton happens at other places too.
>
> I am currently making the same test program using OPAL 2.2.3 (this should be the latest stable release) and for now i do not have those problems. I will post the result when i found out something
>
>
>
> -----------------------------------------------------------------
> Survivor BG. Оцеляването продължава на www.survivor.btv.bg 

Re: [OpenSIPStack] Using OpenSBC

[Joegen E. B. <jo...@pl...>]

Hi David,

Good to hear you have it working.  We do have some known issues with 
some UA's in media proxy mode.  First, can I have more details on your 
setup?   Is opensbc located behind a NATted network?  How bout the IPBX 
and the UA's?   What UA's are involved (manufacture, firmware version, 
etc).   If possible, would you be able to get me some ethereal dumps 
from both the UAC and the UAS (both SIP and RTP)?

Note:  You can disable media proxy mode for public ip's by setting 
m_MediaProxyIfPrivate = FALSE to TRUE in B2BUAConnection constructor.

Joegen


David Craigon wrote:
> Thanks very much for your excellent advice. I've got SIP acting as a
> back to back user agent. Alas, though, I've got no sound! Is there
> anything to do to make the RTP/media go through OpenSBC?
>
> David
>
>   
>> -----Original Message-----
>> From: ope...@li... 
>> [mailto:ope...@li...] On 
>> Behalf Of jo...@pl...
>> Sent: 25 October 2006 18:41
>> To: ope...@li...
>> Subject: Re: [OpenSIPStack] Using OpenSBC
>>
>> inline ...
>>
>> ----- Original Message -----
>> From: "David Craigon" <dav...@gr...>
>> To: <ope...@li...>
>> Sent: Thursday, October 26, 2006 12:23 AM
>> Subject: [OpenSIPStack] Using OpenSBC
>>
>>
>>     
>>> Hello everyone,
>>>    I am trying to work out how to use OpenSBC as a back to back user
>>> agent. I've got it all installed and running, and can 
>>>       
>> configure it via
>>     
>>> port 9999. I'm trying to configure the following simple set up.
>>>
>>>
>>> |--------------------|         x|----------|y   z|--------|
>>> | Naff software SIP  |----------| Open SBC |-----| IP PBX |
>>> | phone              |          |----------|     |--------|
>>> |--------------------|
>>>
>>> X and y are IP addresses.
>>>
>>> What I want to do first is get my SIP software phone to 
>>>       
>> ring the IP PBX
>>     
>>> via OpenSBC. The idea is that you would ring a number on 
>>>       
>> the phone (say
>>     
>>> 1234), which would be translated to 1234@x. This call would 
>>>       
>> then be sent
>>     
>>> to the IP PBX, hopefully as 1234@z.
>>>
>>> So far, I've managed to configure my softphone to register 
>>>       
>> with OpenSBC.
>>     
>>> I have the configuration options on the web menu:
>>>
>>> ------
>>> OpenSBC Configuration Page
>>>       
>> This is where you set Log Level for for pwlib as well as 
>> Admin account and 
>> passwrod
>>
>>     
>>> OpenSBC General Parameters
>>>       
>> This contains general params for opensbc such as sip logging, default 
>> timeout, thread pool size etc.
>>
>>     
>>> Local Domain Accounts
>>>       
>> Local domain accounts allows you to configure user accounts 
>> or the registrar
>>
>>     
>>> OpenSBC Routes
>>>
>>>       
>> This is where you set the B2BUA routes.
>>
>>     
>>> Relay Routes
>>>       
>> This is where you set the routes to be releayed to remote a 
>> remote domain.
>>
>>     
>>> Upper Registration Routes
>>>       
>> Allows OpenSBc o hijack contacts for registration bound to a 
>> remote domain
>>
>>     
>>> ------
>>>
>>> What do I need to configure to make my set up work? I'm 
>>>       
>> guessing that I
>>     
>>> need to put in an OpenSBC route, or a relay route, to route 
>>>       
>> calls from
>>     
>>> the softphone to the PBX. What's the difference between an 
>>>       
>> OpenSBC route
>>     
>>> and a relay route? What syntax do the rules take? You 
>>>       
>> supply the example
>>     
>>> : [sip:*@example.opensipstack.org:*] sip:p1.opensipstack.org:5060,
>>> sip:p2.opensipstack.org:5060. What does this mean?
>>>       
>> You need to put the routes in OpenSBC route if the To URI of 
>> your INVITE 
>> resolves to the IP Address of OpenSBC either using A Record 
>> or SRV Record 
>> DNS lookup.   For example, if OpenSBC resides on 
>> 192.168.0.10, when it 
>> receives an incoming invite, it checks the to URI if it resolves to 
>> 192.168.0.10.  If it does, it would considered a local domain 
>> call and would 
>> use the OpenSBC Route for routing this call.  If its not in 
>> the static 
>> routes, OpenSBc would then check the registration table.
>>
>> In cases where the To URI is not resolving to the OpenSBC 
>> address, it would 
>> assume that you are attempting to relay a message to a remote domain. 
>> OpenSBC, in this case would go into pure stateful proxy mode 
>> and use the 
>> Relay Routes to route the call.  In cases where relay route 
>> does not contain 
>> an entry for the call, it would try to resolve the final 
>> destination using 
>> the to URI or using the Route Set if present.
>>
>> As for the syntax, let me give you an example.   Let us say I 
>> would want to 
>> route calls to 1234 to a certain gateway at 
>> gw01.somedomain.com.  Assume 
>> openSBC is hosted in a box with A-Record set to 
>> mysip.domain.com.  You route 
>> entry in this case would be:
>>
>>  [sip:12...@my...*] sip:12...@gw...
>>
>> The first element enclosed in [] is the wildcard matching 
>> filter.   Upon 
>> receipt of an Invite, OpenSBC would try to match the To URI using all 
>> available filters.  In this case, To: "Alice" 
>> 12...@my...:5060 
>> would match this filter.   The "*" means anycall bound for 
>> 12...@ms... regardless of the port.  the next 
>> element in the route 
>> entry would be the destination address.    When the call is 
>> finally routed, 
>> OpenSBc would rewrite the startline URI with the address you 
>> provided here.
>>
>> (Inbound Invite)
>> INVITE sip:12...@my... SIP/2.0
>> From: sip:the...@my...;tag=mytag-123456
>> To: "Alice" <sip:12...@my...>
>> .
>> .
>> .
>>
>> (Outbound Invite)
>> INVITE sip:gw01.somedomain.com SIP/2.0
>> From: sip:the...@my...;tag=mytag-123456
>> To: "Alice" <sip:12...@my...>
>> .
>> .
>> .
>>
>> Hope this helps.
>>
>>
>> Joegen
>>
>>
>>
>>
>>
>>     
>>> Thanks for any help you can give to this total beginner,
>>>
>>> David
>>>
>>>
>>>       
>> --------------------------------------------------------------
>> -----------
>>     
>>> Using Tomcat but need to do more? Need to support web 
>>>       
>> services, security?
>>     
>>> Get stuff done quickly with pre-integrated technology to 
>>>       
>> make your job 
>>     
>>> easier
>>> Download IBM WebSphere Application Server v.1.0.1 based on 
>>>       
>> Apache Geronimo
>>     
>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&
>> dat=121642
>>     
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel 
>>>       
>> --------------------------------------------------------------
>> -----------
>> Using Tomcat but need to do more? Need to support web 
>> services, security?
>> Get stuff done quickly with pre-integrated technology to make 
>> your job easier
>> Download IBM WebSphere Application Server v.1.0.1 based on 
>> Apache Geronimo
>> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&
>> dat=121642
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>     
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

Re: [OpenSIPStack] Using OpenSBC

[David C. <dav...@gr...>]

Thanks very much for your excellent advice. I've got SIP acting as a
back to back user agent. Alas, though, I've got no sound! Is there
anything to do to make the RTP/media go through OpenSBC?

David

> -----Original Message-----
> From: ope...@li...=20
> [mailto:ope...@li...] On=20
> Behalf Of jo...@pl...
> Sent: 25 October 2006 18:41
> To: ope...@li...
> Subject: Re: [OpenSIPStack] Using OpenSBC
>=20
> inline ...
>=20
> ----- Original Message -----
> From: "David Craigon" <dav...@gr...>
> To: <ope...@li...>
> Sent: Thursday, October 26, 2006 12:23 AM
> Subject: [OpenSIPStack] Using OpenSBC
>=20
>=20
> > Hello everyone,
> >    I am trying to work out how to use OpenSBC as a back to back user
> > agent. I've got it all installed and running, and can=20
> configure it via
> > port 9999. I'm trying to configure the following simple set up.
> >
> >
> > |--------------------|         x|----------|y   z|--------|
> > | Naff software SIP  |----------| Open SBC |-----| IP PBX |
> > | phone              |          |----------|     |--------|
> > |--------------------|
> >
> > X and y are IP addresses.
> >
> > What I want to do first is get my SIP software phone to=20
> ring the IP PBX
> > via OpenSBC. The idea is that you would ring a number on=20
> the phone (say
> > 1234), which would be translated to 1234@x. This call would=20
> then be sent
> > to the IP PBX, hopefully as 1234@z.
> >
> > So far, I've managed to configure my softphone to register=20
> with OpenSBC.
> > I have the configuration options on the web menu:
> >
> > ------
> > OpenSBC Configuration Page
>=20
> This is where you set Log Level for for pwlib as well as=20
> Admin account and=20
> passwrod
>=20
> >
> > OpenSBC General Parameters
>=20
> This contains general params for opensbc such as sip logging, default=20
> timeout, thread pool size etc.
>=20
> >
> > Local Domain Accounts
>=20
> Local domain accounts allows you to configure user accounts=20
> or the registrar
>=20
> >
> > OpenSBC Routes
> >
>=20
> This is where you set the B2BUA routes.
>=20
> > Relay Routes
> This is where you set the routes to be releayed to remote a=20
> remote domain.
>=20
> >
> > Upper Registration Routes
> Allows OpenSBc o hijack contacts for registration bound to a=20
> remote domain
>=20
> > ------
> >
> > What do I need to configure to make my set up work? I'm=20
> guessing that I
> > need to put in an OpenSBC route, or a relay route, to route=20
> calls from
> > the softphone to the PBX. What's the difference between an=20
> OpenSBC route
> > and a relay route? What syntax do the rules take? You=20
> supply the example
> > : [sip:*@example.opensipstack.org:*] sip:p1.opensipstack.org:5060,
> > sip:p2.opensipstack.org:5060. What does this mean?
>=20
>=20
> You need to put the routes in OpenSBC route if the To URI of=20
> your INVITE=20
> resolves to the IP Address of OpenSBC either using A Record=20
> or SRV Record=20
> DNS lookup.   For example, if OpenSBC resides on=20
> 192.168.0.10, when it=20
> receives an incoming invite, it checks the to URI if it resolves to=20
> 192.168.0.10.  If it does, it would considered a local domain=20
> call and would=20
> use the OpenSBC Route for routing this call.  If its not in=20
> the static=20
> routes, OpenSBc would then check the registration table.
>=20
> In cases where the To URI is not resolving to the OpenSBC=20
> address, it would=20
> assume that you are attempting to relay a message to a remote domain.=20
> OpenSBC, in this case would go into pure stateful proxy mode=20
> and use the=20
> Relay Routes to route the call.  In cases where relay route=20
> does not contain=20
> an entry for the call, it would try to resolve the final=20
> destination using=20
> the to URI or using the Route Set if present.
>=20
> As for the syntax, let me give you an example.   Let us say I=20
> would want to=20
> route calls to 1234 to a certain gateway at=20
> gw01.somedomain.com.  Assume=20
> openSBC is hosted in a box with A-Record set to=20
> mysip.domain.com.  You route=20
> entry in this case would be:
>=20
>  [sip:12...@my...*] sip:12...@gw...
>=20
> The first element enclosed in [] is the wildcard matching=20
> filter.   Upon=20
> receipt of an Invite, OpenSBC would try to match the To URI using all=20
> available filters.  In this case, To: "Alice"=20
> 12...@my...:5060=20
> would match this filter.   The "*" means anycall bound for=20
> 12...@ms... regardless of the port.  the next=20
> element in the route=20
> entry would be the destination address.    When the call is=20
> finally routed,=20
> OpenSBc would rewrite the startline URI with the address you=20
> provided here.
>=20
> (Inbound Invite)
> INVITE sip:12...@my... SIP/2.0
> From: sip:the...@my...;tag=3Dmytag-123456
> To: "Alice" <sip:12...@my...>
> .
> .
> .
>=20
> (Outbound Invite)
> INVITE sip:gw01.somedomain.com SIP/2.0
> From: sip:the...@my...;tag=3Dmytag-123456
> To: "Alice" <sip:12...@my...>
> .
> .
> .
>=20
> Hope this helps.
>=20
>=20
> Joegen
>=20
>=20
>=20
>=20
>=20
> >
> > Thanks for any help you can give to this total beginner,
> >
> > David
> >
> >=20
> --------------------------------------------------------------
> -----------
> > Using Tomcat but need to do more? Need to support web=20
> services, security?
> > Get stuff done quickly with pre-integrated technology to=20
> make your job=20
> > easier
> > Download IBM WebSphere Application Server v.1.0.1 based on=20
> Apache Geronimo
> >=20
> http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&
> dat=3D121642
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel=20
>=20
>=20
> --------------------------------------------------------------
> -----------
> Using Tomcat but need to do more? Need to support web=20
> services, security?
> Get stuff done quickly with pre-integrated technology to make=20
> your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on=20
> Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&
> dat=3D121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>=20

Re: [OpenSIPStack] MessengerManager Exception

[Borislav P. <bor...@ab...>]

Hi,
I am no blaming anyone :) , as i said in my first post i am still a c++ n=
ewbie so, probably the prolem is in code. But I saw that thread in the ne=
t: http://www.openh323.org/pipermail/openh323/Week-of-Mon-20060123/076248=
.html and decided that my problem is related.
I can send you the code tomorrow (after about 10 hours).

 >-------- =CE=F0=E8=E3=E8=ED=E0=EB=ED=EE =EF=E8=F1=EC=EE --------
 >=CE=F2:  <jo...@pl...>
 >=CE=F2=ED=EE=F1=ED=EE: Re: [OpenSIPStack] MessengerManager Exception
 >=C4=EE: <ope...@li...>
 >=C8=E7=EF=F0=E0=F2=E5=ED=EE =ED=E0: =D1=F0=FF=E4=E0, 2006, =CE=EA=F2=EE=
=EC=E2=F0=E8 25 21:29:15 GMT+03:00
 >----------------------------------
 >
 >Hi Borislav,
 >
 >Please send me a copy of your revised MessengerManager project files.  =
 I'll=20
 >take a look at it when I get a chance.    Please send off list if the f=
ile=20
 >is large.   Lets not put the blame on Opal.  I'm sure the openh323 guys=
=20
 >would have never allowed a bug in Opal destruction if its there.  ;-)
 >
 >Joegen
 >
 >----- Original Message -----=20
 >From: "Borislav Pechenyashki" <bor...@ab...>
 >To: <ope...@li...>
 >Sent: Thursday, October 26, 2006 1:57 AM
 >Subject: Re: [OpenSIPStack] MessengerManager Exception
 >
 >
 >> Hi, i using MessengerManager outside the ATLSIP, because I want to ma=
ke a=20
 >> dll wrapper instead of activeX control. I have removed all the atl re=
lated=20
 >> stuff, and I use the MessengerManager just as a descendent of OPAlMan=
ager.=20
 >> I think that the exceptions I receive on destruction of the Messenger=
 are=20
 >> cause by some thread issues in the OpalManager itself.
 >> I have found that others have also reported an exception in OpalManag=
er's=20
 >> destruction under Windows so I will experiment with some other OPAL=20
 >> version.
 >>
 >>
 >> >  You might have already found out that ATLSIP is emant to be used a=
s an=20
 >> > ActiveX Control.  MessengerManager is designed to work together wit=
h the=20
 >> > upper layer COM interfaces.  Using it outside of those interfaces m=
ay=20
 >> > lead you to all sorts of library misbehavior.  If I find time, I wo=
uld=20
 >> > make MessagerManager a separate library so that none ATL applicatio=
n may=20
 >> > ling to it directly.  For now, you may want to dig deeper as to the=
=20
 >> > cause of the exception and post your results here.  Maybe I or some=
one=20
 >> > else would pitch in an advise.
 >> >
 >> >  Joegen
 >> >
 >> >  >-----Original Message-----
 >> >  >From: Borislav Pechenyashki [mailto:bor...@ab...]
 >> >  >Sent: Tuesday, October 24, 2006 02:25 PM
 >> >  >To: ope...@li...
 >> >  >Subject: [OpenSIPStack] MessengerManager Exception
 >> >  >
 >> >  >Hi, i am still trying ATLSIP and I noticed that an exception is t=
hrown=20
 >> > when MessengerManager is destroyed(); (The exception appers only if=
=20
 >> > InitializeSIP is called. If i do not call InitializeSIP then there =
is no=20
 >> > exception)
 >> >  >
 >> >  >The exception:
 >> >  >First-chance exception at 0x7c812a5b in PchOSSTest.exe: 0x0000071=
A:=20
 >> > The remote procedure call was cancelled.
 >> >  >
 >> >  >I made a console test program ( below ):
 >> >  >when i enter '0' the sipstack starts destroying and then throws a=
n=20
 >> > exception somewhere. (The messenger class is modified a little - th=
e=20
 >> > atlsip reference and all FIRE_ events are removed)
 >> >  >I am a newbie in C++ so may be I am doing something wrong but i w=
ant=20
 >> > to ask just in case if there is such a known issue.
 >> >  >
 >> >  >#include "stdafx.h"
 >> >  >#include <iostream>
 >> >  >#include "Messenger.h"
 >> >  >#include "windows.h"
 >> >  >
 >> >  >PDECLARE_PROCESS( DummyPProcess,=20
 >> > PProcess,"PchOSSApi","PchOSSApi",MAJOR_VERSION,MINOR_VERSION,BUILD_=
TYPE,BUILD_NUMBER)
 >> >  >void DummyPProcess::Main(){};
 >> >  >static DummyPProcess process; // active for the life of the DLL
 >> >  >
 >> >  >int _tmain(int argc, _TCHAR* argv[])
 >> >  >{
 >> >  > MessengerManager * manager =3D new MessengerManager(0, 0);
 >> >  > manager->InitializeSIP();
 >> >  > char ch;
 >> >  > do
 >> >  > {
 >> >  > cin >> ch;
 >> >  > } while (ch !=3D '0');
 >> >  >
 >> >  > delete manager;
 >> >  >
 >> >  > return 0;
 >> >  >}
 >> >  >
 >> >  >-----------------------------------------------------------------
 >> >  >Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=
=E6=E0=E2=E0 =ED=E0 www.survivor.btv.bg
 >> >  >
 >> >=20
 >> >  >-----------------------------------------------------------------=
--------
 >> >  >Using Tomcat but need to do more? Need to support web services,=20
 >> > security?
 >> >  >Get stuff done quickly with pre-integrated technology to make you=
r job=20
 >> > easier
 >> >  >Download IBM WebSphere Application Server v.1.0.1 based on Apache=
=20
 >> > Geronimo
 >> >=20
 >> >  >http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D2630=
57&dat=3D121642
 >> >  >_______________________________________________
 >> >  >opensipstack-devel mailing list
 >> >  >ope...@li...
 >> >  >https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
 >> >  >
 >> >
 >>
 >> -----------------------------------------------------------------
 >> Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=E6=E0=
=E2=E0 =ED=E0 www.survivor.btv.bg
 >>
 >> ---------------------------------------------------------------------=
----
 >> Using Tomcat but need to do more? Need to support web services, secur=
ity?
 >> Get stuff done quickly with pre-integrated technology to make your jo=
b=20
 >> easier
 >> Download IBM WebSphere Application Server v.1.0.1 based on Apache Ger=
onimo
 >> http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&d=
at=3D121642
 >> _______________________________________________
 >> opensipstack-devel mailing list
 >> ope...@li...
 >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
 >>=20
 >
 >
 >-----------------------------------------------------------------------=
--
 >Using Tomcat but need to do more? Need to support web services, securit=
y?
 >Get stuff done quickly with pre-integrated technology to make your job =
easier
 >Download IBM WebSphere Application Server v.1.0.1 based on Apache Geron=
imo
 >http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=
=3D121642
 >_______________________________________________
 >opensipstack-devel mailing list
 >ope...@li...
 >https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
 >

-----------------------------------------------------------------
Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=E6=E0=E2=
=E0 =ED=E0 www.survivor.btv.bg=20

Re: [OpenSIPStack] MessengerManager Exception

[<jo...@pl...>]

Hi Borislav,

Please send me a copy of your revised MessengerManager project files.   I'll 
take a look at it when I get a chance.    Please send off list if the file 
is large.   Lets not put the blame on Opal.  I'm sure the openh323 guys 
would have never allowed a bug in Opal destruction if its there.  ;-)

Joegen

----- Original Message ----- 
From: "Borislav Pechenyashki" <bor...@ab...>
To: <ope...@li...>
Sent: Thursday, October 26, 2006 1:57 AM
Subject: Re: [OpenSIPStack] MessengerManager Exception


> Hi, i using MessengerManager outside the ATLSIP, because I want to make a 
> dll wrapper instead of activeX control. I have removed all the atl related 
> stuff, and I use the MessengerManager just as a descendent of OPAlManager. 
> I think that the exceptions I receive on destruction of the Messenger are 
> cause by some thread issues in the OpalManager itself.
> I have found that others have also reported an exception in OpalManager's 
> destruction under Windows so I will experiment with some other OPAL 
> version.
>
>
> >  You might have already found out that ATLSIP is emant to be used as an 
> > ActiveX Control.  MessengerManager is designed to work together with the 
> > upper layer COM interfaces.  Using it outside of those interfaces may 
> > lead you to all sorts of library misbehavior.  If I find time, I would 
> > make MessagerManager a separate library so that none ATL application may 
> > ling to it directly.  For now, you may want to dig deeper as to the 
> > cause of the exception and post your results here.  Maybe I or someone 
> > else would pitch in an advise.
> >
> >  Joegen
> >
> >  >-----Original Message-----
> >  >From: Borislav Pechenyashki [mailto:bor...@ab...]
> >  >Sent: Tuesday, October 24, 2006 02:25 PM
> >  >To: ope...@li...
> >  >Subject: [OpenSIPStack] MessengerManager Exception
> >  >
> >  >Hi, i am still trying ATLSIP and I noticed that an exception is thrown 
> > when MessengerManager is destroyed(); (The exception appers only if 
> > InitializeSIP is called. If i do not call InitializeSIP then there is no 
> > exception)
> >  >
> >  >The exception:
> >  >First-chance exception at 0x7c812a5b in PchOSSTest.exe: 0x0000071A: 
> > The remote procedure call was cancelled.
> >  >
> >  >I made a console test program ( below ):
> >  >when i enter '0' the sipstack starts destroying and then throws an 
> > exception somewhere. (The messenger class is modified a little - the 
> > atlsip reference and all FIRE_ events are removed)
> >  >I am a newbie in C++ so may be I am doing something wrong but i want 
> > to ask just in case if there is such a known issue.
> >  >
> >  >#include "stdafx.h"
> >  >#include <iostream>
> >  >#include "Messenger.h"
> >  >#include "windows.h"
> >  >
> >  >PDECLARE_PROCESS( DummyPProcess, 
> > PProcess,"PchOSSApi","PchOSSApi",MAJOR_VERSION,MINOR_VERSION,BUILD_TYPE,BUILD_NUMBER)
> >  >void DummyPProcess::Main(){};
> >  >static DummyPProcess process; // active for the life of the DLL
> >  >
> >  >int _tmain(int argc, _TCHAR* argv[])
> >  >{
> >  > MessengerManager * manager = new MessengerManager(0, 0);
> >  > manager->InitializeSIP();
> >  > char ch;
> >  > do
> >  > {
> >  > cin >> ch;
> >  > } while (ch != '0');
> >  >
> >  > delete manager;
> >  >
> >  > return 0;
> >  >}
> >  >
> >  >-----------------------------------------------------------------
> >  >Survivor BG. Оцеляването продължава на www.survivor.btv.bg
> >  >
> > 
> >  >-------------------------------------------------------------------------
> >  >Using Tomcat but need to do more? Need to support web services, 
> > security?
> >  >Get stuff done quickly with pre-integrated technology to make your job 
> > easier
> >  >Download IBM WebSphere Application Server v.1.0.1 based on Apache 
> > Geronimo
> > 
> >  >http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> >  >_______________________________________________
> >  >opensipstack-devel mailing list
> >  >ope...@li...
> >  >https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >  >
> >
>
> -----------------------------------------------------------------
> Survivor BG. Оцеляването продължава на www.survivor.btv.bg
>
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job 
> easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> 

Re: [OpenSIPStack] MessengerManager Exception

[Borislav P. <bor...@ab...>]

Hi, i using MessengerManager outside the ATLSIP, because I want to make a=
 dll wrapper instead of activeX control. I have removed all the atl relat=
ed stuff, and I use the MessengerManager just as a descendent of OPAlMana=
ger. I think that the exceptions I receive on destruction of the Messenge=
r are cause by some thread issues in the OpalManager itself.
I have found that others have also reported an exception in OpalManager's=
 destruction under Windows so I will experiment with some other OPAL vers=
ion.=20


 >  You might have already found out that ATLSIP is emant to be used as a=
n ActiveX Control.  MessengerManager is designed to work together with th=
e upper layer COM interfaces.  Using it outside of those interfaces may l=
ead you to all sorts of library misbehavior.  If I find time, I would mak=
e MessagerManager a separate library so that none ATL application may lin=
g to it directly.  For now, you may want to dig deeper as to the cause of=
 the exception and post your results here.  Maybe I or someone else would=
 pitch in an advise.
 >
 >  Joegen
 >
 >  >-----Original Message-----
 >  >From: Borislav Pechenyashki [mailto:bor...@ab...]
 >  >Sent: Tuesday, October 24, 2006 02:25 PM
 >  >To: ope...@li...
 >  >Subject: [OpenSIPStack] MessengerManager Exception
 >  >
 >  >Hi, i am still trying ATLSIP and I noticed that an exception is thro=
wn when MessengerManager is destroyed(); (The exception appers only if In=
itializeSIP is called. If i do not call InitializeSIP then there is no ex=
ception)
 >  >
 >  >The exception:
 >  >First-chance exception at 0x7c812a5b in PchOSSTest.exe: 0x0000071A: =
The remote procedure call was cancelled.
 >  >
 >  >I made a console test program ( below ):
 >  >when i enter '0' the sipstack starts destroying and then throws an e=
xception somewhere. (The messenger class is modified a little - the atlsi=
p reference and all FIRE_ events are removed)
 >  >I am a newbie in C++ so may be I am doing something wrong but i want=
 to ask just in case if there is such a known issue.
 >  >
 >  >#include "stdafx.h"
 >  >#include <iostream>
 >  >#include "Messenger.h"
 >  >#include "windows.h"
 >  >
 >  >PDECLARE_PROCESS( DummyPProcess, PProcess,"PchOSSApi","PchOSSApi",MA=
JOR_VERSION,MINOR_VERSION,BUILD_TYPE,BUILD_NUMBER)
 >  >void DummyPProcess::Main(){};
 >  >static DummyPProcess process; // active for the life of the DLL
 >  >
 >  >int _tmain(int argc, _TCHAR* argv[])
 >  >{
 >  > MessengerManager * manager =3D new MessengerManager(0, 0);
 >  > manager->InitializeSIP();=20
 >  > char ch;
 >  > do
 >  > {
 >  > cin >> ch;=20
 >  > } while (ch !=3D '0');
 >  >
 >  > delete manager;
 >  >=20
 >  > return 0;
 >  >}
 >  >
 >  >-----------------------------------------------------------------
 >  >Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=E6=
=E0=E2=E0 =ED=E0 www.survivor.btv.bg=20
 >  >
 >  >--------------------------------------------------------------------=
-----
 >  >Using Tomcat but need to do more? Need to support web services, secu=
rity?
 >  >Get stuff done quickly with pre-integrated technology to make your j=
ob easier
 >  >Download IBM WebSphere Application Server v.1.0.1 based on Apache Ge=
ronimo
 >  >http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&=
dat=3D121642
 >  >_______________________________________________
 >  >opensipstack-devel mailing list
 >  >ope...@li...
 >  >https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
 >  >
 >

-----------------------------------------------------------------
Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =EF=F0=EE=E4=FA=EB=E6=E0=E2=
=E0 =ED=E0 www.survivor.btv.bg=20

Re: [OpenSIPStack] MessengerManager Exception

[<jo...@pl...>]

Resending this one ...



  Hi Borislav,

  You might have already found out that ATLSIP is emant to be used as an =
ActiveX Control.  MessengerManager is designed to work together with the =
upper layer COM interfaces.  Using it outside of those interfaces may =
lead you to all sorts of library misbehavior.  If I find time, I would =
make MessagerManager a separate library so that none ATL application may =
ling to it directly.  For now, you may want to dig deeper as to the =
cause of the exception and post your results here.  Maybe I or someone =
else would pitch in an advise.

  Joegen

  >-----Original Message-----
  >From: Borislav Pechenyashki [mailto:bor...@ab...]
  >Sent: Tuesday, October 24, 2006 02:25 PM
  >To: ope...@li...
  >Subject: [OpenSIPStack] MessengerManager Exception
  >
  >Hi, i am still trying ATLSIP and I noticed that an exception is =
thrown when MessengerManager is destroyed(); (The exception appers only =
if InitializeSIP is called. If i do not call InitializeSIP then there is =
no exception)
  >
  >The exception:
  >First-chance exception at 0x7c812a5b in PchOSSTest.exe: 0x0000071A: =
The remote procedure call was cancelled.
  >
  >I made a console test program ( below ):
  >when i enter '0' the sipstack starts destroying and then throws an =
exception somewhere. (The messenger class is modified a little - the =
atlsip reference and all FIRE_ events are removed)
  >I am a newbie in C++ so may be I am doing something wrong but i want =
to ask just in case if there is such a known issue.
  >
  >#include "stdafx.h"
  >#include <iostream>
  >#include "Messenger.h"
  >#include "windows.h"
  >
  >PDECLARE_PROCESS( DummyPProcess, =
PProcess,"PchOSSApi","PchOSSApi",MAJOR_VERSION,MINOR_VERSION,BUILD_TYPE,B=
UILD_NUMBER)
  >void DummyPProcess::Main(){};
  >static DummyPProcess process; // active for the life of the DLL
  >
  >int _tmain(int argc, _TCHAR* argv[])
  >{
  > MessengerManager * manager =3D new MessengerManager(0, 0);
  > manager->InitializeSIP();=20
  > char ch;
  > do
  > {
  > cin >> ch;=20
  > } while (ch !=3D '0');
  >
  > delete manager;
  >=20
  > return 0;
  >}
  >
  >-----------------------------------------------------------------
  >Survivor BG. =CE=F6=E5=EB=FF=E2=E0=ED=E5=F2=EE =
=EF=F0=EE=E4=FA=EB=E6=E0=E2=E0 =ED=E0 www.survivor.btv.bg=20
  >
  =
>------------------------------------------------------------------------=
-
  >Using Tomcat but need to do more? Need to support web services, =
security?
  >Get stuff done quickly with pre-integrated technology to make your =
job easier
  >Download IBM WebSphere Application Server v.1.0.1 based on Apache =
Geronimo
  =
>http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D=
121642
  >_______________________________________________
  >opensipstack-devel mailing list
  >ope...@li...
  >https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
  >