opensipstack-devel

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[dial2world d. <dia...@ho...>]

Dear Ilion.,

Thanks for the prompt response...

We did test welltech USB phones..There was no problem with the audio 
interopability but the keypad functionality is not happening..

Given to understand that so far no thorough tests were made...Well its a 
request... in the current competitive market ATL SIP SF also need to be 
compatible with major  USB phone vendors  ...

As you are aware  all the major SFs like Xlite, net2phone, eyebeam SJ phone 
and many others are compatible with major usb vendors ...

How to go about the same ?? Can we participate in the test for 
interopability?? We can request the vendor for any assistance..

We have currently sourced outthe below  usb phones from TCL ;-

http://www.tclcomm.cn/product_detail.asp?TypeID=12&Pid=10149
http://www.tclcomm.cn/product_detail.asp?TypeID=12&Pid=10128

Thanks in advance

regards
Salish



>From: "Ilian Jeri C. Pinzon" <ip...@so...>
>Reply-To: ope...@li...
>To: fo...@op...,ope...@li...
>Subject: Re: [OpenSIPStack] Problem with OnOutgoingCallConnected
>Date: Mon, 09 Jul 2007 13:48:46 +0800
>
>Hi,
>
>Nope. ATLSIP doesn't have this functionality *yet*.
>
>- Ilian
>
>tomach wrote:
> > Thanks it worked :)
> >
> >
> > Hmmm does ATLSIP have a functionality to play wave file to the remote 
>subscriber (outgoing call) through RTP protocol?
> >
> > 
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>
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Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[Ilian J. C. P. <ip...@so...>]

Hi,

Nope. ATLSIP doesn't have this functionality *yet*.

- Ilian

tomach wrote:
> Thanks it worked :)
>
>
> Hmmm does ATLSIP have a functionality to play wave file to the remote subscriber (outgoing call) through RTP protocol?
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] compatible usb phones with OSS phone dialer built/developed through ATLSIP

[Ilian J. C. P. <ip...@so...>]

Hi,

No thorough tests were done on ATLSIP to determine what USB phones are 
compatible with it. But as long as the USB phone is detected by the OS 
as a sound device, it should work.

Regards,
Ilian

dial2world dl2 wrote:
> Hi there .,
>
> Would appreciate if somebody can let me know which vendors usb phones and 
> usb gateways are compatible with softphone OSS phone  dialer built/developed 
> through ATLSIP 1.0.2 API/SDK ??
>
> thank you for your co-operation in advance
>
> Regards
> Amita
>
> _________________________________________________________________
> Millions of profiles with photos. Search now @ Shaadi.com 
> http://ss1.richmedia.in/recurl.asp?pid=108
>
>
> -------------------------------------------------------------------------
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> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
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> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] OpenSIPStack: Problem

[Joegen E. B. <joe...@gm...>]

*/Windows.h missing/* is a complile error commonly reported in the 
mailing list. This may happen in Visual Stdio 2005 Express edition of 
visual C++. Windows.h goes with the Platform SDK and is available via a 
separate installation here 
http://msdn.microsoft.com/vstudio/express/visualc/usingpsdk/.

Arif V. Kadiwal wrote:
> Dear Developers/Staff,
>
> I am using OpenSIPStack 1.1.5, according to your web documentation
> under section Compiling In Windows I was following steps to compile
> OpenSIPStack solution in VS 8. During the compilation time it give me
> error in file contain.h saying it is unable to find file windows.h the
> detail error is given below I got lots of error with this file
> (windows.h is not found)
>
> c:\documents and settings\student6\desktop\open source
> sipstack\opensipstack-1.1.5\opensipstack-1.1.5-rc1\include\ptlib\msos/ptlib/contain.h(290)
> : fatal error C1083: Cannot open include file: 'windows.h': No such
> file or directory
> pxmlrpc.cxx
>
>
> from the above it seems there is no windows.h file in OpenSIPStakc
> folder hierarchy and I also looked thoroughly and i didnt find
> windows.h file in OpenSIPStack folder tree. Can please help me to
> rectify this problem. I would appreciate your quick soon in this
> regard.
>
>
> Arif
>
> Research Assistant
> Electrical and Computer Engineering
> Concordia Unviersity
> Montreal, Canada
> www.concordia.ca
>
> -------------------------------------------------------------------------
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>   

[OpenSIPStack] OpenSIPStack: Problem

[Arif V. K. <msg...@gm...>]

Dear Developers/Staff,

I am using OpenSIPStack 1.1.5, according to your web documentation
under section Compiling In Windows I was following steps to compile
OpenSIPStack solution in VS 8. During the compilation time it give me
error in file contain.h saying it is unable to find file windows.h the
detail error is given below I got lots of error with this file
(windows.h is not found)

c:\documents and settings\student6\desktop\open source
sipstack\opensipstack-1.1.5\opensipstack-1.1.5-rc1\include\ptlib\msos/ptlib/contain.h(290)
: fatal error C1083: Cannot open include file: 'windows.h': No such
file or directory
pxmlrpc.cxx


from the above it seems there is no windows.h file in OpenSIPStakc
folder hierarchy and I also looked thoroughly and i didnt find
windows.h file in OpenSIPStack folder tree. Can please help me to
rectify this problem. I would appreciate your quick soon in this
regard.


Arif

Research Assistant
Electrical and Computer Engineering
Concordia Unviersity
Montreal, Canada
www.concordia.ca

Re: [OpenSIPStack] Error when ending application

[Joegen E. B. <joe...@gm...>]

Hi Vincent,

Before we proceed further in figuring out the cause of your problems, 
please upgrade your code using the latest CVS HEAD.   There have been a 
lot of changes to ATLSIP and the Softphone API since 1.1.5.



Joegen



Vincent GROMFELD wrote:
> Hello,
>
>  
>
> I'm currently trying your sip stack. All works pretty well but I noticed two
> annoying things. I'm working on Windows Vista using Visual Studio 2005 to
> build and debug. I'm using the latest stable release of the stack : 1.1.5.
>
>  
>
> First, if I call someone, and then me or him hang-up, the line is still
> opened on his or my side. If I hang-up, his softphone will not see that I've
> hung up and It he hangs up, the softphone sample provided with ATL SIP will
> not seen that he has hung up. Do you have already seen this problem before ?
> This is not a big problem but it will better if someone hang up that both
> see that line is down.
>
>  
>
> The second thing happened at closure and is really annoying. When
> application exit, I see in Visual Studio 2005 debugger window an exception
> telling  "INVALID STRING BINDING" and nothing more happened; the application
> freezes. This occurs only after performing a call. If I start the
> application without calling anybody, I have no problems. I have the problem
> in debug and even in release mode. I cannot see where the error come, the
> only thing I found is that this error is raised by RPC calls.
>
>  
>
> Do you have any solution for these problems ? 
>
> Thanks for your help.
>
>  
>
> Vincent GROMFELD 
>
> -------------------------------------------------------------------------
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> Download DB2 Express C - the FREE version of DB2 express and take
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>
>   

[OpenSIPStack] Error when ending application

[Vincent G. <vin...@sm...>]

Hello,

 

I'm currently trying your sip stack. All works pretty well but I noticed two
annoying things. I'm working on Windows Vista using Visual Studio 2005 to
build and debug. I'm using the latest stable release of the stack : 1.1.5.

 

First, if I call someone, and then me or him hang-up, the line is still
opened on his or my side. If I hang-up, his softphone will not see that I've
hung up and It he hangs up, the softphone sample provided with ATL SIP will
not seen that he has hung up. Do you have already seen this problem before ?
This is not a big problem but it will better if someone hang up that both
see that line is down.

 

The second thing happened at closure and is really annoying. When
application exit, I see in Visual Studio 2005 debugger window an exception
telling  "INVALID STRING BINDING" and nothing more happened; the application
freezes. This occurs only after performing a call. If I start the
application without calling anybody, I have no problems. I have the problem
in debug and even in release mode. I cannot see where the error come, the
only thing I found is that this error is raised by RPC calls.

 

Do you have any solution for these problems ? 

Thanks for your help.

 

Vincent GROMFELD 

[OpenSIPStack] compatible usb phones with OSS phone dialer built/developed through ATLSIP

[dial2world d. <dia...@ho...>]

Hi there .,

Would appreciate if somebody can let me know which vendors usb phones and 
usb gateways are compatible with softphone OSS phone  dialer built/developed 
through ATLSIP 1.0.2 API/SDK ??

thank you for your co-operation in advance

Regards
Amita

_________________________________________________________________
Millions of profiles with photos. Search now @ Shaadi.com 
http://ss1.richmedia.in/recurl.asp?pid=108

Re: [OpenSIPStack] [OpenSipStack] Problem registering with OSSPhone (+questions)

[Joegen E. B. <joe...@gm...>]

sebastian pastor wrote:
> Hi all,
>
>
>
> i've downladed the latest versions of opensipstack and atlsip using cvs. i
> also downloaded windows installer for OSSPhone.
>
>
>
> I have a very basic problem when registering: it sends the register to the
> configured registrar server (OK) but in from and to headers it will populate
> also that registar regardless what i have put as user_name or phone_number.
>
>
>
> See logs:
>
> Hey when i installed it again it was magically solved!!!!!
>
>
>
> Anyway, I notice in mailing lists people asking questions with higher
> complexity within SIP protocol so i assume they are registered properly and
> can make basic calls. Then, this issue must be (or have en) related with my
> configuration. This makes me wonder some things:
>
>
>
> 1.Does ATLSIP library have to be compiled in my computer so that OSSPhone
> works OK? Or the installer automatically does it when creates dinamic
> libraries?
>   


Not necessarily.    What you encountered is a difference in version.  
The softphone installer is very old.  Installing from CVS is always the 
recommended path.

>
>
> 2.Can OSSPhone be used as a SIP phone by itself (with any account in iptel,
> sylantro or any server...)?

YES.  Most definitely. 

>  Or it has to work together with SBC?
>   

No, it doesn't require OpenSBC.

>
>
> 3.If i want to modify some behaviour of this softphone what code should i
> edit? ATLSIP is just a library but not a normal one... it creates dlls!! so
> where is the "pure" code where i can find the basic behaviour of the phone
> (you know, e.g: a main() where the initial GUI is launched or the calls to
> the different methods defined in opensipstack)
>
>
>   

I think Ilian is best to answer this.  From what I understand, .NET 
Form  and MFC applications have different entry point than int main().



>
>
> Thanks in advance
>
>
>
>                         -Sebastián-
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

Re: [OpenSIPStack] SipUri Parse Error

[Joegen E. B. <joe...@gm...>]

Hi Gustavo,

Yes, I see the fault in the code.   Patch is again in CVS.  Thanks for 
reporting the bug.

Joegen


Gustavo Curetti wrote:
> Hi Joegen:
>  
> I have checked the patch, but the comparation:
>  
> BOOL hasEnclosure = begin != P_MAX_INDEX && end != P_MAX_INDEX;
>  
> is still wrong if the "begin" value is set like:
>
> PINDEX begin = value.Find( '<' ) + 1;
>  
> I suggest:
>  
> PINDEX begin = value.Find( '<' );
> PINDEX end = value.Find( '>' );
> BOOL hasEnclosure = begin != P_MAX_INDEX && end != P_MAX_INDEX; 
> PString uri;
>
> if( hasEnclosure )
> {
>  PINDEX rightOffSet = value.GetLength() - end;
>  PINDEX length = value.GetLength() - begin *- 1* - rightOffSet;
>  uri = value.Mid( begin *+ 1*, length ).Trim();
> }
>  
> And in the code:
>  
> PString tBuff = uri;
>  
> PINDEX atIndex = *value*.FindLast( '@' <mailto:%27@%27> );
>  
> if( atIndex != P_MAX_INDEX )
>     tBuff = uri.Mid( atIndex + 1 );
>
> The value of atIndex is obtained from value instead of from uri.
> I suggest:
>  
> PINDEX atIndex = *uri*.FindLast( '@' <mailto:%27@%27> );
>  
> Or:
>  
> PString tBuff = value;
> PINDEX atIndex = value.FindLast( '@' );
>  
> if( atIndex != P_MAX_INDEX )
> {
>     if( !hasEnclosure )
>     {
>         tBuff = value.Mid( atIndex + 1 );
>     } else {
>         atIndex = uri.FindLast( '@' );
>         tBuff = uri.Mid( atIndex + 1 );
>     }
> }
>  
> Regards,
> Gustavo
>
> ------------------------------------------------------------------------
>
> > Date: Wed, 4 Jul 2007 14:57:16 +0800
> > To: cur...@gm...
> > From: joe...@gm...
> > CC: ope...@li...
> > Subject: Re: [OpenSIPStack] SipUri Parse Error
> >
> > I have committed a patch for this bug. Thanks for reporting it.
> >
> >
> > Gustavo Curetti wrote:
> > > Joegen:
> > >
> > > When an uri like:
> > >
> > > <sip:782706@192.168.0.1:5060;transport=udp;user=phone>
> > >
> > > is parsed by SIPURI & SIPURI::operator=(const MimeHeader & header)
> > >
> > > the '>' is kept in the param user (user=phone>).
> > >
> > > Then the result uri is:
> > >
> > > sip:782706@192.168.0.1:5060;transport=udp;user=phone>
> > >
> > > When this uri is parsed again by SIPURI & SIPURI::operator=(const
> > > MimeHeader & header),
> > >
> > > the uri is detected like it has enclosure, but the uri is not 
> extracted.
> > >
> > > I made a few changes to correct the second error:
> > >
> > > PINDEX begin = value.Find( '<' ) + 1; --> PINDEX begin = value.Find(
> > > '<' ); (to be compared with P_MAX_INDEX)
> > > PINDEX end = value.Find( '>' );
> > > BOOL hasEnclosure = begin != P_MAX_INDEX && end != P_MAX_INDEX;
> > > PString uri;
> > >
> > > if( hasEnclosure )
> > > {
> > > PINDEX rightOffSet = value.GetLength() - end;
> > > PINDEX length = value.GetLength() - begin - rightOffSet; --> PINDEX
> > > length = value.GetLength() - begin - 1 - rightOffSet;
> > > uri = value.Mid( begin, length ).Trim(); --> uri = value.Mid( begin +
> > > 1, length ).Trim();
> > > }
> > >
> > > And to correct the first error:
> > >
> > > PStringArray t1, t2;
> > > PString tBuff = value;
> > > PINDEX atIndex = value.FindLast( '@' );
> > >
> > > if( atIndex != P_MAX_INDEX )
> > > tBuff = value.Mid( atIndex + 1 );
> > > |
> > > V
> > > PStringArray t1, t2;
> > > PString tBuff = value;
> > > PINDEX atIndex = value.FindLast( '@' );
> > >
> > > if( atIndex != P_MAX_INDEX )
> > > {
> > > if( !hasEnclosure )
> > > {
> > > tBuff = value.Mid( atIndex + 1 );
> > > } else {
> > > atIndex = uri.FindLast( '@' );
> > > tBuff = uri.Mid( atIndex + 1 );
> > > }
> > > }
> > >
> > > Please check if these changes are ok.
> > >
> > > Regards,
> > > Gustavo
> > >
> > > 
> ------------------------------------------------------------------------
> > > Comunícate al instante con Windows Live Messenger Windows Live
> > > Messenger
> > > 
> <http://imagine-msn.com/messenger/launch80/default.aspx?locale=es-ar&source=joinmsncom/messenger>
> >
> >
> > 
> -------------------------------------------------------------------------
> > This SF.net email is sponsored by DB2 Express
> > Download DB2 Express C - the FREE version of DB2 express and take
> > control of your XML. No limits. Just data. Click to get it now.
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> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
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[OpenSIPStack] [OpenSipStack] Problem registering with OSSPhone (+questions)

[sebastian p. <seb...@gm...>]

Hi all,



i've downladed the latest versions of opensipstack and atlsip using cvs. i
also downloaded windows installer for OSSPhone.



I have a very basic problem when registering: it sends the register to the
configured registrar server (OK) but in from and to headers it will populat=
e
also that registar regardless what i have put as user_name or phone_number.



See logs:

Hey when i installed it again it was magically solved!!!!!



Anyway, I notice in mailing lists people asking questions with higher
complexity within SIP protocol so i assume they are registered properly and
can make basic calls. Then, this issue must be (or have en) related with my
configuration. This makes me wonder some things:



1.Does ATLSIP library have to be compiled in my computer so that OSSPhone
works OK? Or the installer automatically does it when creates dinamic
libraries?



2.Can OSSPhone be used as a SIP phone by itself (with any account in iptel,
sylantro or any server...)? Or it has to work together with SBC?



3.If i want to modify some behaviour of this softphone what code should i
edit? ATLSIP is just a library but not a normal one... it creates dlls!! so
where is the "pure" code where i can find the basic behaviour of the phone
(you know, e.g: a main() where the initial GUI is launched or the calls to
the different methods defined in opensipstack)





Thanks in advance



                        -Sebasti=E1n-

Re: [OpenSIPStack] SipUri Parse Error

[Gustavo C. <cur...@ho...>]

Hi Joegen: I have checked the patch, but the comparation: BOOL hasEnclosure=
 =3D begin !=3D P_MAX_INDEX && end !=3D P_MAX_INDEX; is still wrong if the =
"begin" value is set like:PINDEX begin =3D value.Find( '<' ) + 1; I suggest=
: PINDEX begin =3D value.Find( '<' );PINDEX end =3D value.Find( '>' );BOOL =
hasEnclosure =3D begin !=3D P_MAX_INDEX && end !=3D P_MAX_INDEX; PString ur=
i;if( hasEnclosure ){ PINDEX rightOffSet =3D value.GetLength() - end; PINDE=
X length =3D value.GetLength() - begin - 1 - rightOffSet; uri =3D value.Mid=
( begin + 1, length ).Trim();} And in the code: PString tBuff =3D uri; PIND=
EX atIndex =3D value.FindLast( '@' ); if( atIndex !=3D P_MAX_INDEX )    tBu=
ff =3D uri.Mid( atIndex + 1 );The value of atIndex is obtained from value i=
nstead of from uri.I suggest: PINDEX atIndex =3D uri.FindLast( '@' ); Or: P=
String tBuff =3D value;PINDEX atIndex =3D value.FindLast( '@' ); if( atInde=
x !=3D P_MAX_INDEX ){    if( !hasEnclosure )    {        tBuff =3D value.Mi=
d( atIndex + 1 );    } else {        atIndex =3D uri.FindLast( '@' );      =
  tBuff =3D uri.Mid( atIndex + 1 );    }} Regards,Gustavo

> Date: Wed, 4 Jul 2007 14:57:16 +0800> To: cur...@gm...> From=
: joe...@gm...> CC: ope...@li...> Su=
bject: Re: [OpenSIPStack] SipUri Parse Error> > I have committed a patch fo=
r this bug. Thanks for reporting it.> > > Gustavo Curetti wrote:> > Joegen:=
> >> > When an uri like:> >> > <sip:782706@192.168.0.1:5060;transport=3Dudp=
;user=3Dphone>> >> > is parsed by SIPURI & SIPURI::operator=3D(const MimeHe=
ader & header)> >> > the '>' is kept in the param user (user=3Dphone>). > >=
> > Then the result uri is:> >> > sip:782706@192.168.0.1:5060;transport=3Du=
dp;user=3Dphone>> >> > When this uri is parsed again by SIPURI & SIPURI::op=
erator=3D(const > > MimeHeader & header),> >> > the uri is detected like it=
 has enclosure, but the uri is not extracted.> >> > I made a few changes to=
 correct the second error:> >> > PINDEX begin =3D value.Find( '<' ) + 1; --=
> PINDEX begin =3D value.Find( > > '<' ); (to be compared with P_MAX_INDEX)=
> > PINDEX end =3D value.Find( '>' );> > BOOL hasEnclosure =3D begin !=3D P=
_MAX_INDEX && end !=3D P_MAX_INDEX; > > PString uri;> >> > if( hasEnclosure=
 )> > {> > PINDEX rightOffSet =3D value.GetLength() - end;> > PINDEX length=
 =3D value.GetLength() - begin - rightOffSet; --> PINDEX > > length =3D val=
ue.GetLength() - begin - 1 - rightOffSet;> > uri =3D value.Mid( begin, leng=
th ).Trim(); --> uri =3D value.Mid( begin + > > 1, length ).Trim();> > }> >=
 > > And to correct the first error:> > > > PStringArray t1, t2;> > PString=
 tBuff =3D value;> > PINDEX atIndex =3D value.FindLast( '@' );> > > > if( a=
tIndex !=3D P_MAX_INDEX )> > tBuff =3D value.Mid( atIndex + 1 );> > |> > V>=
 > PStringArray t1, t2;> > PString tBuff =3D value;> > PINDEX atIndex =3D v=
alue.FindLast( '@' );> >> > if( atIndex !=3D P_MAX_INDEX )> > {> > if( !has=
Enclosure )> > {> > tBuff =3D value.Mid( atIndex + 1 );> > } else {> > atIn=
dex =3D uri.FindLast( '@' );> > tBuff =3D uri.Mid( atIndex + 1 );> > }> > }=
> >> > Please check if these changes are ok.> > > > Regards,> > Gustavo> >>=
 > ------------------------------------------------------------------------=
> > Comun=EDcate al instante con Windows Live Messenger Windows Live > > Me=
ssenger > > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=
=3Des-ar&source=3Djoinmsncom/messenger>> > > ------------------------------=
-------------------------------------------> This SF.net email is sponsored=
 by DB2 Express> Download DB2 Express C - the FREE version of DB2 express a=
nd take> control of your XML. No limits. Just data. Click to get it now.> h=
ttp://sourceforge.net/powerbar/db2/> ______________________________________=
_________> opensipstack-devel mailing list> ope...@li...urce=
forge.net> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

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Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Thanks it worked :)


Hmmm does ATLSIP have a functionality to play wave file to the remote subscriber (outgoing call) through RTP protocol?

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[Ilian J. C. P. <ip...@so...>]

Hi,

Ringing is performed by the following functions:

Ringing for incoming calls is invoked by: 
SoftPhoneSIPEndPoint::OnSetupIncomingCall(...)

Ringing for outgoing calls is invoked by: 
SoftPhoneSIPEndPoint::OnAlerting(...)

The actual ringing is done by SoftPhoneInterface::PlayRingBackTone( 
PThread &, INT param ).

The ringing is stopped when SoftPhoneInterface::StopRingBackTone() is 
called.

Depending on your approach, you can edit any of these functions to play 
your custom ringing.

- Ilian

tomach wrote:
> Ehh I celebrate to fast :)
>
> When I change ringing sound then also sound (tone) that I hear in ATLSoftphone changes. I am interested to hear different sound when somebody call to me and hear different when i call to remote subscriber. Is it possible at all?
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy

[Joegen E. B. <joe...@gm...>]

Gustavo Curetti wrote:
> Hi Joegen:
>  
> I don't have a call sesion, i have a proxy session. The cancel is sent 
> to the m_DialogPeerAddress for a proxy session. I have added the 
> following code:
>  
> void ProxySession::OnDNSFailOver(
> SIPDNSFailOver & failOverEvent
> )
> {
>  SIPMessage msg = failOverEvent.GetMessage();
>  if( msg.IsInvite() )
>  {
>   SIPURI targetURI;
>   msg.GetRequestURI(targetURI);
>   SetDialogPeerAddress( targetURI.AsString() );
>  }
> }

I have patched CVS accordingly.


>  
> Then the cancel is sent to the actual target of the ICT transaction, 
> but if the actual target is not answering, the ICT continue trying 
> with the actual target and with the followings. The ICT transaction 
> must stop trying. Could you give some directions, please?

I think this is a compliant behavior.   Don't you want the UA to know 
that the UAS is actually down?  I have checked 3261 again and paragraph 
indicates that a proxy MAY send a response to the CANCEL and generate a 
new transaction towards the UAS.   This is not a MUST and is left to the 
implementors to decide on.  I am open to discuss this further if you 
think it is a better approach to handle extra state in the proxy for 
cancelling transaction.   My main reason for doing this is to make proxy 
transactions light and I would favor to let it stay that way.

    9.2 Server Behavior

       The CANCEL method requests that the TU at the server side cancel a
       pending transaction.  The TU determines the transaction to be
       cancelled by taking the CANCEL request, and then assuming that the
       request method is anything but CANCEL or ACK and applying the
       transaction matching procedures of Section 17.2.3.  The matching
       transaction is the one to be cancelled.

       The processing of a CANCEL request at a server depends on the type of
       server.  A stateless proxy will forward it, a stateful proxy might
       respond to it and generate some CANCEL requests of its own, and a UAS
       will respond to it.  See Section 16.10 for proxy treatment of CANCEL.




>
> Thanks for your help.
> Gustavo.
>  
> ------------------------------------------------------------------------
>
> > Date: Tue, 26 Jun 2007 15:23:39 +0800
> > From: jb...@so...
> > To: jb...@so...; ope...@li...
> > CC: cur...@gm...
> > Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy
> >
> > Hi Gustavo,
> >
> > I have committed a fix of this. It's very simple actually. I just
> > propagated a SIPDNSFailOver stack event to the call session and made
> > sure that m_CurrentUACInvite is replaced. Can you check if this is a
> > safe change? Thanks.
> >
> > Joegen
> >
> >
> > Joegen E. Baclor wrote:
> > > Hmmmn thats TRUE. Lemme dig further. I'll let you know once a fix is
> > > available.
> > >
> > >
> > > Gustavo Curetti wrote:
> > >
> > >> Joegen:
> > >>
> > >> The fix works fine. But i have the following problem: When i hang up
> > >> and send a Cancel to OpenSBC, the OpenSBC respond with
> > >> Code487_RequestCancelled but the Cancel is not been send to the 
> actual
> > >> target, the Cancel is just routed again.
> > >>
> > >> Thanks for your help.
> > >>
> > >> Gustavo
> > >>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>
> > >>> Date: Wed, 13 Jun 2007 22:15:14 +0800
> > >>> To: cur...@gm...; 
> ope...@li...
> > >>> Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy
> > >>> From: joe...@gm...
> > >>>
> > >>> I've just checked in a fix for this in CVS.
> > >>>
> > >>> Gustavo Curetti wrote:
> > >>>
> > >>>> Joegen,
> > >>>>
> > >>>> I tried the DoDNSFailover's code but i have the following behavior:
> > >>>>
> > >>>> When DoDNSFailover() is called for the first time 
> m_FailOverAttempts
> > >>>> is increased from 0 to 1.
> > >>>>
> > >>>> Then the via is changed (+ "-") and with this change when
> > >>>> FindTransactionAndAddEvent() is called a new transaction is 
> created.
> > >>>> In SIPTransaction::SIPTransaction() m_FailOverAttempts is 
> initialized
> > >>>> to 0. Then when DoDNSFailover() is called again, the same target
> > >>>>
> > >> is used.
> > >>
> > >>>> Gustavo
> > >>>>
> > >>>>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>> From: cur...@ho...
> > >>>> To: jb...@so...
> > >>>> Subject: RE: [OpenSIPStack] FW: OpenSBC as Forking Proxy
> > >>>> Date: Wed, 30 May 2007 14:43:32 +0200
> > >>>>
> > >>>> Joegen,
> > >>>>
> > >>>> Thanks a lot. I will try to change the behavior.
> > >>>>
> > >>>> Gustavo
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>> Date: Fri, 18 May 2007 08:15:20 +0800
> > >>>>> From: jb...@so...
> > >>>>> To: cur...@gm...;
> > >>>>>
> > >>>> ope...@li...
> > >>>>
> > >>>>> Subject: Re: [OpenSIPStack] FW: OpenSBC as Forking Proxy
> > >>>>>
> > >>>>> Gustavo,
> > >>>>>
> > >>>>> Sorry, I forgot to get back to you. If you have the latest CVS 
> head
> > >>>>> code, check out BOOL SIPTransaction::DoDNSFailover() in
> > >>>>> SIPTransaction.cxx. I have committed this a few days ago to
> > >>>>> demonstrate fail-over by forking using DNS/SRV records. New
> > >>>>> transactions are created by calling
> > >>>>>
> > >>>> FindTransactionAndAddEvent(). You
> > >>>>
> > >>>>> can just change its behavior a bit and get the fail-over routes
> > >>>>> somewhere instead of DNS/SRV queries.
> > >>>>>
> > >>>>> Joegen
> > >>>>>
> > >>>>>
> > >>>>> Gustavo Curetti wrote:
> > >>>>>
> > >>>>>> Joegen:
> > >>>>>>
> > >>>>>> I don't understand how to create a new client transaction when
> > >>>>>>
> > >>>> the
> > >>>>
> > >>>>>> first invite fail in the FSM layer. Could you give some
> > >>>>>>
> > >>>> directions,
> > >>>>
> > >>>>>> please?
> > >>>>>>
> > >>>>>> Thanks for your help.
> > >>>>>>
> > >>>>>> Gustavo.
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>> From: cur...@ho...
> > >>>>>> To: jb...@so...
> > >>>>>> Subject: RE: [OpenSIPStack] OpenSBC as Forking Proxy
> > >>>>>> Date: Mon, 14 May 2007 16:18:57 +0200
> > >>>>>>
> > >>>>>> Joegen:
> > >>>>>>
> > >>>>>> Thanks for your help. Do you suggest to do the serial forking in
> > >>>>>> FSM layer with a custom header?.Must each new try create a new
> > >>>>>> client transaction?
> > >>>>>>
> > >>>>>> Thanks
> > >>>>>> Gustavo
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>> Date: Thu, 10 May 2007 13:28:29 +0800
> > >>>>>>> From: jb...@so...
> > >>>>>>> To: cur...@gm...
> > >>>>>>> Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
> > >>>>>>>
> > >>>>>>> Gustavo,
> > >>>>>>>
> > >>>>>>> This will be a bit tricky. It is not as simple as spawning an
> > >>>>>>>
> > >>>>>> outbound
> > >>>>>>
> > >>>>>>> invite. There should be a clean mechanism to clone
> > >>>>>>>
> > >>>> transactions and
> > >>>>
> > >>>>>>> this is not present in the FSM currently. Forking should be done
> > >>>>>>>
> > >>>>>> in the
> > >>>>>>
> > >>>>>>> FSM layer, not in the UACore layer. I will see what I can do to
> > >>>>>>>
> > >>>>>> help
> > >>>>>>
> > >>>>>>> you. I will let you know when I have something you can use
> > >>>>>>>
> > >>>>>> cleanly to
> > >>>>>>
> > >>>>>>> fork your calls. Perhaps over the weekend, but that isn't a
> > >>>>>>>
> > >>>> promise.
> > >>>>
> > >>>>>>> Joegen
> > >>>>>>>
> > >>>>>>> Gustavo Curetti wrote:
> > >>>>>>>
> > >>>>>>>> Joegen:
> > >>>>>>>>
> > >>>>>>>> What I want to do is a very simple sequential search. When one
> > >>>>>>>> destination don't answer or reject the call I want the OpenSBC
> > >>>>>>>>
> > >>>>>> try
> > >>>>>>
> > >>>>>>>> another.
> > >>>>>>>>
> > >>>>>>>> I made the following changes in the code for timer B
> > >>>>>>>>
> > >>>>>> expiration just
> > >>>>>>
> > >>>>>>>> for do some tests:
> > >>>>>>>>
> > >>>>>>>> void ProxySessionManager::OnTimerExpire(
> > >>>>>>>> SIPTimerExpire & timerEvent,
> > >>>>>>>> SIPSession * session
> > >>>>>>>> )
> > >>>>>>>> {
> > >>>>>>>> if( session != NULL )
> > >>>>>>>> {
> > >>>>>>>> LOG_IF_DEBUG( LogWarning(), "*** TIMER EXPIRATION *** for SIP
> > >>>>>>>> Session " << session->GetSessionId() );
> > >>>>>>>> if( timerEvent.GetTimer() ==
> > >>>>>>>>
> > >>>> SIPTransactions::SIPTimerEvent::B)
> > >>>>
> > >>>>>>>> {
> > >>>>>>>> SIPMessage msg = ((ProxySession
> > >>>>>>>>
> > >>>> *)session)->GetOriginalInvite();
> > >>>>
> > >>>>>>>> session->EnqueueSessionEvent( new SIPSessionEvent(
> > >>>>>>>>
> > >>>> *session, 1,
> > >>>>
> > >>>>>>>> msg ) );
> > >>>>>>>> }
> > >>>>>>>> session->OnTimerExpire( timerEvent );
> > >>>>>>>> }
> > >>>>>>>> }
> > >>>>>>>>
> > >>>>>>>> and
> > >>>>>>>>
> > >>>>>>>> void ProxySession::OnTimerExpire(
> > >>>>>>>> SIPTimerExpire & timerEvent
> > >>>>>>>> )
> > >>>>>>>> {
> > >>>>>>>> GCREF( "SIPSession::OnTimerExpire" );
> > >>>>>>>> if( timerEvent.GetTimer() ==
> > >>>>>>>>
> > >>>> SIPTransactions::SIPTimerEvent::B ||
> > >>>>
> > >>>>>>>> timerEvent.GetTimer() == SIPTransactions::SIPTimerEvent::F )
> > >>>>>>>> {
> > >>>>>>>> ///this is an ICT timeout
> > >>>>>>>> SIPMessage timeout;
> > >>>>>>>> GetCurrentUASRequest().CreateResponse( timeout,
> > >>>>>>>> SIPMessage::Code480_TemporarilyNotAvailable );
> > >>>>>>>> SendRequest( timeout );
> > >>>>>>>> }
> > >>>>>>>>
> > >>>>>>>> //Destroy();
> > >>>>>>>> }
> > >>>>>>>>
> > >>>>>>>> With these changes and a relay route:
> > >>>>>>>>
> > >>>>>>>> [sip:*@192.168.0.207:*] sip:192.168.0.1:5060,
> > >>>>>>>>
> > >>>>>> sip:192.168.0.60:5060
> > >>>>>>
> > >>>>>>>> the OpenSBC made the second invite successfully. But what i
> > >>>>>>>>
> > >>>>>> really
> > >>>>>>
> > >>>>>>>> want is to use some custom headers with a list destination
> > >>>>>>>>
> > >>>>>> addresses
> > >>>>>>
> > >>>>>>>> instead of the relay routes and to do the same in case of a
> > >>>>>>>>
> > >>>>>> reject. Do
> > >>>>>>
> > >>>>>>>> you have any suggestions?
> > >>>>>>>>
> > >>>>>>>> Other question: Can i have two active ICT for a session?
> > >>>>>>>>
> > >>>>>> Because in
> > >>>>>>
> > >>>>>>>> the case of reject, I must start a new ICT for trying the next
> > >>>>>>>> destination but canceling throw the first ICT at the same
> > >>>>>>>>
> > >>>> time.
> > >>>>
> > >>>>>>>> Thanks for your help.
> > >>>>>>>>
> > >>>>>>>> Gustavo
> > >>>>>>>>
> > >>>>>>>>
> > >>>>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>>>> Date: Sat, 5 May 2007 13:50:44 +0800
> > >>>>>>>>> From: jb...@so...
> > >>>>>>>>> To: cur...@gm...;
> > >>>>>>>>>
> > >>>>>> ope...@li...
> > >>>>>>
> > >>>>>>>>> Subject: Re: [OpenSIPStack] OpenSBC as Forking Proxy
> > >>>>>>>>>
> > >>>>>>>>> Gustavo,
> > >>>>>>>>>
> > >>>>>>>>> Forking is not supported yet in OpenSBC.
> > >>>>>>>>>
> > >>>>>>>>> Joegen
> > >>>>>>>>>
> > >>>>>>>>> Gustavo Curetti wrote:
> > >>>>>>>>>
> > >>>>>>>>>> Hi Joegen:
> > >>>>>>>>>>
> > >>>>>>>>>> I want to use the OpenSBC as a Forking Proxy. I want
> > >>>>>>>>>>
> > >>>> that the
> > >>>>
> > >>>>>>>>>> OpenSBC try the differents Relays Routes one by one. Could
> > >>>>>>>>>>
> > >>>>>> you give
> > >>>>>>
> > >>>>>>>>>> some directions, please?
> > >>>>>>>>>>
> > >>>>>>>>>> Thanks for your help.
> > >>>>>>>>>>
> > >>>>>>>>>> Gustavo.
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>>>>> Descubre Live.com - tu propia página de inicio,
> > >>>>>>>>>>
> > >>>>>> personalizada para
> > >>>>>>
> > >>>>>>>> ver
> > >>>>>>>>
> > >>>>>>>>>> rápidamente todo lo que te interesa en un mismo sitio.
> > >>>>>>>>>>
> > >>>>>> todo en el
> > >>>>>>
> > >>>>>>>>>> mismo sitio. <http://www.live.com/getstarted>
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>>>>>
> > >> 
> -------------------------------------------------------------------------
> > >>
> > >>>>>>>>>> This SF.net email is sponsored by DB2 Express
> > >>>>>>>>>> Download DB2 Express C - the FREE version of DB2 express
> > >>>>>>>>>>
> > >>>>>> and take
> > >>>>>>
> > >>>>>>>>>> control of your XML. No limits. Just data. Click to get it
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> > >>>>>>
> > >>>>>>>>>> http://sourceforge.net/powerbar/db2/
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>>>>> _______________________________________________
> > >>>>>>>>>> opensipstack-devel mailing list
> > >>>>>>>>>> ope...@li...
> > >>>>>>>>>>
> > >>>>>>>>>>
> > >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >>>>>>
> > >>>>>>>>>
> > >>>>>>>>>
> > >> 
> -------------------------------------------------------------------------
> > >>
> > >>>>>>>>> This SF.net email is sponsored by DB2 Express
> > >>>>>>>>> Download DB2 Express C - the FREE version of DB2 express and
> > >>>>>>>>>
> > >>>>>> take
> > >>>>>>
> > >>>>>>>>> control of your XML. No limits. Just data. Click to get
> > >>>>>>>>>
> > >>>> it now.
> > >>>>
> > >>>>>>>>> http://sourceforge.net/powerbar/db2/
> > >>>>>>>>> _______________________________________________
> > >>>>>>>>> opensipstack-devel mailing list
> > >>>>>>>>> ope...@li...
> > >>>>>>>>>
> > >>>>>>>>>
> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >>>>
> > >>>>>>>>
> > >>>>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>>>> Se uno de los primeros en probar Windows Live Mail. Windows
> > >>>>>>>>
> > >>>>>> Live Mail.
> > >>>>>>
> > >> 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > >>
> > >>>>>>
> > >>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>> Se uno de los primeros en probar Windows Live Mail. Windows Live
> > >>>>>> Mail.
> > >>>>>>
> > >>>>>>
> > >> 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > >>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>>> Se uno de los primeros en probar Windows Live Mail. Windows
> > >>>>>>
> > >>>> Live Mail.
> > >>>>
> > >> 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > >>
> > >>>>>
> > >>>>>
> > >> 
> -------------------------------------------------------------------------
> > >>
> > >>>>> This SF.net email is sponsored by DB2 Express
> > >>>>> Download DB2 Express C - the FREE version of DB2 express and take
> > >>>>> control of your XML. No limits. Just data. Click to get it now.
> > >>>>> http://sourceforge.net/powerbar/db2/
> > >>>>> _______________________________________________
> > >>>>> opensipstack-devel mailing list
> > >>>>> ope...@li...
> > >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >>>>>
> > >>>>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>> Se uno de los primeros en probar Windows Live Mail. Windows Live
> > >>>> Mail.
> > >>>>
> > >>>>
> > >> 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > >>
> > >>>>
> > >>>>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>> Comunícate al instante con Windows Live Messenger Windows Live
> > >>>> Messenger
> > >>>>
> > >>>>
> > >> 
> <http://imagine-msn.com/messenger/launch80/default.aspx?locale=es-ar&source=joinmsncom/messenger> 
>
> > >>
> > >>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>>
> > >> 
> -------------------------------------------------------------------------
> > >>
> > >>>> This SF.net email is sponsored by DB2 Express
> > >>>> Download DB2 Express C - the FREE version of DB2 express and take
> > >>>> control of your XML. No limits. Just data. Click to get it now.
> > >>>> http://sourceforge.net/powerbar/db2/
> > >>>>
> > >>>>
> > >> 
> ------------------------------------------------------------------------
> > >>
> > >>>> _______________________________________________
> > >>>> opensipstack-devel mailing list
> > >>>> ope...@li...
> > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >>>>
> > >>>>
> > >>> --
> > >>> Joegen E. Baclor
> > >>> CTO - Solegy LLC
> > >>>
> > >>> Email: joegen @ solegy.com
> > >>> Main: +1 212 801 2504
> > >>> Fax: +1 347 438 3072
> > >>> Manila: +63 2 747 3460
> > >>> Mobile: +63 918 411 9064
> > >>>
> > >>> 121 Varick St., Suite 201
> > >>> NY, NY 10013
> > >>>
> > >>> SOLEGY LLC
> > >>> http://www.solegy.com
> > >>> Solutions to Fit Your Strategy
> > >>>
> > >>>
> > >>>
> > >> 
> ------------------------------------------------------------------------
> > >> Se uno de los primeros en probar Windows Live Mail. Windows Live 
> Mail.
> > >> 
> <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d>
> > >>
> > >
> > >
> > >
> > > 
> -------------------------------------------------------------------------
> > > This SF.net email is sponsored by DB2 Express
> > > Download DB2 Express C - the FREE version of DB2 express and take
> > > control of your XML. No limits. Just data. Click to get it now.
> > > http://sourceforge.net/powerbar/db2/
> > > _______________________________________________
> > > opensipstack-devel mailing list
> > > ope...@li...
> > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> > >
> > >
> >
> >
> > 
> -------------------------------------------------------------------------
> > This SF.net email is sponsored by DB2 Express
> > Download DB2 Express C - the FREE version of DB2 express and take
> > control of your XML. No limits. Just data. Click to get it now.
> > http://sourceforge.net/powerbar/db2/
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
> ------------------------------------------------------------------------
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Re: [OpenSIPStack] SipUri Parse Error

[Joegen E. B. <joe...@gm...>]

I have committed a patch for this bug.  Thanks for reporting it.


Gustavo Curetti wrote:
> Joegen:
>
> When an uri like:
>
> <sip:782706@192.168.0.1:5060;transport=udp;user=phone>
>
> is parsed by SIPURI & SIPURI::operator=(const MimeHeader & header)
>
> the '>' is kept in the param user (user=phone>). 
>
> Then the result uri is:
>
> sip:782706@192.168.0.1:5060;transport=udp;user=phone>
>
> When this uri is parsed again by SIPURI & SIPURI::operator=(const 
> MimeHeader & header),
>
> the uri is detected like it has enclosure, but the uri is not extracted.
>
> I made a few changes to correct the second error:
>
> PINDEX begin = value.Find( '<' ) + 1; --> PINDEX begin = value.Find( 
> '<' ); (to be compared with P_MAX_INDEX)
> PINDEX end = value.Find( '>' );
> BOOL hasEnclosure = begin != P_MAX_INDEX && end != P_MAX_INDEX; 
> PString uri;
>
> if( hasEnclosure )
> {
>  PINDEX rightOffSet = value.GetLength() - end;
>  PINDEX length = value.GetLength() - begin - rightOffSet; --> PINDEX 
> length = value.GetLength() - begin - 1 - rightOffSet;
>  uri = value.Mid( begin, length ).Trim(); --> uri = value.Mid( begin + 
> 1, length ).Trim();
> }
>  
> And to correct the first error:
>  
> PStringArray t1, t2;
> PString tBuff = value;
> PINDEX atIndex = value.FindLast( '@' );
>  
> if( atIndex != P_MAX_INDEX )
>  tBuff = value.Mid( atIndex + 1 );
> |
> V
> PStringArray t1, t2;
> PString tBuff = value;
> PINDEX atIndex = value.FindLast( '@' );
>
> if( atIndex != P_MAX_INDEX )
> {
>  if( !hasEnclosure )
>  {
>   tBuff = value.Mid( atIndex + 1 );
>  } else {
>   atIndex = uri.FindLast( '@' );
>   tBuff = uri.Mid( atIndex + 1 );
>  }
> }
>
> Please check if these changes are ok.
>  
> Regards,
> Gustavo
>
> ------------------------------------------------------------------------
> Comunícate al instante con Windows Live Messenger Windows Live 
> Messenger 
> <http://imagine-msn.com/messenger/launch80/default.aspx?locale=es-ar&source=joinmsncom/messenger>

Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy

[Gustavo C. <cur...@ho...>]

Hi Joegen:
=20
I don't have a call sesion, i have a proxy session. The cancel is sent to t=
he m_DialogPeerAddress for a proxy session. I have added the following code=
:
=20
void ProxySession::OnDNSFailOver(
SIPDNSFailOver & failOverEvent
)
{
 SIPMessage msg =3D failOverEvent.GetMessage();
 if( msg.IsInvite() )
 {
  SIPURI targetURI;
  msg.GetRequestURI(targetURI);
  SetDialogPeerAddress( targetURI.AsString() );
 }
}
=20
Then the cancel is sent to the actual target of the ICT transaction, but if=
 the actual target is not answering, the ICT continue trying with the actua=
l target and with the followings. The ICT transaction must stop trying. Cou=
ld you give some directions, please?
Thanks for your help.Gustavo.=20



> Date: Tue, 26 Jun 2007 15:23:39 +0800> From: jb...@so...> T=
o: jb...@so...; ope...@li...> CC:=
 cur...@gm...> Subject: Re: [OpenSIPStack] FW: FW: OpenSBC as =
Forking Proxy> > Hi Gustavo,> > I have committed a fix of this. It's very s=
imple actually. I just > propagated a SIPDNSFailOver stack event to the cal=
l session and made > sure that m_CurrentUACInvite is replaced. Can you chec=
k if this is a > safe change? Thanks.> > Joegen> > > Joegen E. Baclor wrote=
:> > Hmmmn thats TRUE. Lemme dig further. I'll let you know once a fix is >=
 > available.> >> >> > Gustavo Curetti wrote:> > > >> Joegen:> >> > >> The =
fix works fine. But i have the following problem: When i hang up > >> and s=
end a Cancel to OpenSBC, the OpenSBC respond with > >> Code487_RequestCance=
lled but the Cancel is not been send to the actual > >> target, the Cancel =
is just routed again.> >>> >> Thanks for your help.> >> > >> Gustavo> >>> >=
> ------------------------------------------------------------------------>=
 >>> >> > >>> Date: Wed, 13 Jun 2007 22:15:14 +0800> >>> To: curetti.gustav=
o...@gm...; ope...@li...> >>> Subject: Re: [Op=
enSIPStack] FW: FW: OpenSBC as Forking Proxy> >>> From: joegen.baclor@gmail=
.com> >>>> >>> I've just checked in a fix for this in CVS.> >>>> >>> Gustav=
o Curetti wrote:> >>> > >>>> Joegen,> >>>>> >>>> I tried the DoDNSFailover'=
s code but i have the following behavior:> >>>>> >>>> When DoDNSFailover() =
is called for the first time m_FailOverAttempts> >>>> is increased from 0 t=
o 1.> >>>>> >>>> Then the via is changed (+ "-") and with this change when>=
 >>>> FindTransactionAndAddEvent() is called a new transaction is created.>=
 >>>> In SIPTransaction::SIPTransaction() m_FailOverAttempts is initialized=
> >>>> to 0. Then when DoDNSFailover() is called again, the same target > >=
>>> > >> is used.> >> > >>>> Gustavo> >>>>> >>>>> >>>> > >> ---------------=
---------------------------------------------------------> >> > >>>> From: =
cur...@ho...> >>>> To: jb...@so...> >>>> Subje=
ct: RE: [OpenSIPStack] FW: OpenSBC as Forking Proxy> >>>> Date: Wed, 30 May=
 2007 14:43:32 +0200> >>>>> >>>> Joegen,> >>>>> >>>> Thanks a lot. I will t=
ry to change the behavior.> >>>>> >>>> Gustavo> >>>>> >>>>> >>>>> >>>> > >>=
 ------------------------------------------------------------------------> =
>> > >>>>> Date: Fri, 18 May 2007 08:15:20 +0800> >>>>> From: jbaclor@soleg=
ysystems.com> >>>>> To: cur...@gm...;> >>>>> > >>>> opensipsta=
ck-...@li...> >>>> > >>>>> Subject: Re: [OpenSIPStack] FW=
: OpenSBC as Forking Proxy> >>>>>> >>>>> Gustavo,> >>>>>> >>>>> Sorry, I fo=
rgot to get back to you. If you have the latest CVS head> >>>>> code, check=
 out BOOL SIPTransaction::DoDNSFailover() in> >>>>> SIPTransaction.cxx. I h=
ave committed this a few days ago to> >>>>> demonstrate fail-over by forkin=
g using DNS/SRV records. New> >>>>> transactions are created by calling> >>=
>>> > >>>> FindTransactionAndAddEvent(). You> >>>> > >>>>> can just change =
its behavior a bit and get the fail-over routes> >>>>> somewhere instead of=
 DNS/SRV queries.> >>>>>> >>>>> Joegen> >>>>>> >>>>>> >>>>> Gustavo Curetti=
 wrote:> >>>>> > >>>>>> Joegen:> >>>>>>> >>>>>> I don't understand how to c=
reate a new client transaction when> >>>>>> > >>>> the> >>>> > >>>>>> first=
 invite fail in the FSM layer. Could you give some> >>>>>> > >>>> direction=
s,> >>>> > >>>>>> please?> >>>>>>> >>>>>> Thanks for your help.> >>>>>>> >>=
>>>> Gustavo.> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >> ----------------=
--------------------------------------------------------> >> > >>>>>> From:=
 cur...@ho...> >>>>>> To: jb...@so...> >>>>>> =
Subject: RE: [OpenSIPStack] OpenSBC as Forking Proxy> >>>>>> Date: Mon, 14 =
May 2007 16:18:57 +0200> >>>>>>> >>>>>> Joegen:> >>>>>>> >>>>>> Thanks for =
your help. Do you suggest to do the serial forking in> >>>>>> FSM layer wit=
h a custom header?.Must each new try create a new> >>>>>> client transactio=
n?> >>>>>>> >>>>>> Thanks> >>>>>> Gustavo> >>>>>>> >>>>>>> >>>>>> > >> ----=
--------------------------------------------------------------------> >> > =
>>>>>>> Date: Thu, 10 May 2007 13:28:29 +0800> >>>>>>> From: jbaclor@solegy=
systems.com> >>>>>>> To: cur...@gm...> >>>>>>> Subject: Re: [O=
penSIPStack] OpenSBC as Forking Proxy> >>>>>>>> >>>>>>> Gustavo,> >>>>>>>> =
>>>>>>> This will be a bit tricky. It is not as simple as spawning an> >>>>=
>>> > >>>>>> outbound> >>>>>> > >>>>>>> invite. There should be a clean mec=
hanism to clone> >>>>>>> > >>>> transactions and> >>>> > >>>>>>> this is no=
t present in the FSM currently. Forking should be done> >>>>>>> > >>>>>> in=
 the> >>>>>> > >>>>>>> FSM layer, not in the UACore layer. I will see what =
I can do to> >>>>>>> > >>>>>> help> >>>>>> > >>>>>>> you. I will let you kn=
ow when I have something you can use> >>>>>>> > >>>>>> cleanly to> >>>>>> >=
 >>>>>>> fork your calls. Perhaps over the weekend, but that isn't a> >>>>>=
>> > >>>> promise.> >>>> > >>>>>>> Joegen> >>>>>>>> >>>>>>> Gustavo Curetti=
 wrote:> >>>>>>> > >>>>>>>> Joegen:> >>>>>>>>> >>>>>>>> What I want to do i=
s a very simple sequential search. When one> >>>>>>>> destination don't ans=
wer or reject the call I want the OpenSBC> >>>>>>>> > >>>>>> try> >>>>>> > =
>>>>>>>> another.> >>>>>>>>> >>>>>>>> I made the following changes in the c=
ode for timer B> >>>>>>>> > >>>>>> expiration just> >>>>>> > >>>>>>>> for d=
o some tests:> >>>>>>>>> >>>>>>>> void ProxySessionManager::OnTimerExpire(>=
 >>>>>>>> SIPTimerExpire & timerEvent,> >>>>>>>> SIPSession * session> >>>>=
>>>> )> >>>>>>>> {> >>>>>>>> if( session !=3D NULL )> >>>>>>>> {> >>>>>>>> =
LOG_IF_DEBUG( LogWarning(), "*** TIMER EXPIRATION *** for SIP> >>>>>>>> Ses=
sion " << session->GetSessionId() );> >>>>>>>> if( timerEvent.GetTimer() =
=3D=3D> >>>>>>>> > >>>> SIPTransactions::SIPTimerEvent::B)> >>>> > >>>>>>>>=
 {> >>>>>>>> SIPMessage msg =3D ((ProxySession> >>>>>>>> > >>>> *)session)-=
>GetOriginalInvite();> >>>> > >>>>>>>> session->EnqueueSessionEvent( new SI=
PSessionEvent(> >>>>>>>> > >>>> *session, 1,> >>>> > >>>>>>>> msg ) );> >>>=
>>>>> }> >>>>>>>> session->OnTimerExpire( timerEvent );> >>>>>>>> }> >>>>>>=
>> }> >>>>>>>>> >>>>>>>> and> >>>>>>>>> >>>>>>>> void ProxySession::OnTimer=
Expire(> >>>>>>>> SIPTimerExpire & timerEvent> >>>>>>>> )> >>>>>>>> {> >>>>=
>>>> GCREF( "SIPSession::OnTimerExpire" );> >>>>>>>> if( timerEvent.GetTime=
r() =3D=3D> >>>>>>>> > >>>> SIPTransactions::SIPTimerEvent::B ||> >>>> > >>=
>>>>>> timerEvent.GetTimer() =3D=3D SIPTransactions::SIPTimerEvent::F )> >>=
>>>>>> {> >>>>>>>> ///this is an ICT timeout> >>>>>>>> SIPMessage timeout;>=
 >>>>>>>> GetCurrentUASRequest().CreateResponse( timeout,> >>>>>>>> SIPMess=
age::Code480_TemporarilyNotAvailable );> >>>>>>>> SendRequest( timeout );> =
>>>>>>>> }> >>>>>>>>> >>>>>>>> //Destroy();> >>>>>>>> }> >>>>>>>>> >>>>>>>>=
 With these changes and a relay route:> >>>>>>>>> >>>>>>>> [sip:*@192.168.0=
.207:*] sip:192.168.0.1:5060,> >>>>>>>> > >>>>>> sip:192.168.0.60:5060> >>>=
>>> > >>>>>>>> the OpenSBC made the second invite successfully. But what i>=
 >>>>>>>> > >>>>>> really> >>>>>> > >>>>>>>> want is to use some custom hea=
ders with a list destination> >>>>>>>> > >>>>>> addresses> >>>>>> > >>>>>>>=
> instead of the relay routes and to do the same in case of a> >>>>>>>> > >=
>>>>> reject. Do> >>>>>> > >>>>>>>> you have any suggestions?> >>>>>>>>> >>=
>>>>>> Other question: Can i have two active ICT for a session?> >>>>>>>> >=
 >>>>>> Because in> >>>>>> > >>>>>>>> the case of reject, I must start a ne=
w ICT for trying the next> >>>>>>>> destination but canceling throw the fir=
st ICT at the same> >>>>>>>> > >>>> time.> >>>> > >>>>>>>> Thanks for your =
help.> >>>>>>>>> >>>>>>>> Gustavo> >>>>>>>>> >>>>>>>>> >>>>>>>> > >> ------=
------------------------------------------------------------------> >> > >>=
>>>>>>> Date: Sat, 5 May 2007 13:50:44 +0800> >>>>>>>>> From: jbaclor@soleg=
ysystems.com> >>>>>>>>> To: cur...@gm...;> >>>>>>>>> > >>>>>> =
ope...@li...> >>>>>> > >>>>>>>>> Subject: Re: [=
OpenSIPStack] OpenSBC as Forking Proxy> >>>>>>>>>> >>>>>>>>> Gustavo,> >>>>=
>>>>>> >>>>>>>>> Forking is not supported yet in OpenSBC.> >>>>>>>>>> >>>>>=
>>>> Joegen> >>>>>>>>>> >>>>>>>>> Gustavo Curetti wrote:> >>>>>>>>> > >>>>>=
>>>>> Hi Joegen:> >>>>>>>>>>> >>>>>>>>>> I want to use the OpenSBC as a For=
king Proxy. I want> >>>>>>>>>> > >>>> that the> >>>> > >>>>>>>>>> OpenSBC t=
ry the differents Relays Routes one by one. Could> >>>>>>>>>> > >>>>>> you =
give> >>>>>> > >>>>>>>>>> some directions, please?> >>>>>>>>>>> >>>>>>>>>> =
Thanks for your help.> >>>>>>>>>>> >>>>>>>>>> Gustavo.> >>>>>>>>>>> >>>>>>>=
>>>> >>>>>>>>>> > >> ------------------------------------------------------=
------------------> >> > >>>>>>>>>> Descubre Live.com - tu propia p=E1gina =
de inicio,> >>>>>>>>>> > >>>>>> personalizada para> >>>>>> > >>>>>>>> ver> =
>>>>>>>> > >>>>>>>>>> r=E1pidamente todo lo que te interesa en un mismo sit=
io.> >>>>>>>>>> > >>>>>> todo en el> >>>>>> > >>>>>>>>>> mismo sitio. <http=
://www.live.com/getstarted>> >>>>>>>>>>> >>>>>>>>>> > >> ------------------=
------------------------------------------------------> >> > >>>>>>>>>> > >=
> -------------------------------------------------------------------------=
> >> > >>>>>>>>>> This SF.net email is sponsored by DB2 Express> >>>>>>>>>>=
 Download DB2 Express C - the FREE version of DB2 express> >>>>>>>>>> > >>>=
>>> and take> >>>>>> > >>>>>>>>>> control of your XML. No limits. Just data=
. Click to get it> >>>>>>>>>> > >>>>>> now.> >>>>>> > >>>>>>>>>> http://sou=
rceforge.net/powerbar/db2/> >>>>>>>>>>> >>>>>>>>>> > >> -------------------=
-----------------------------------------------------> >> > >>>>>>>>>> ____=
___________________________________________> >>>>>>>>>> opensipstack-devel =
mailing list> >>>>>>>>>> ope...@li...> >>>>>>>>=
>>> >>>>>>>>>> > >>>>>> https://lists.sourceforge.net/lists/listinfo/opensi=
pstack-devel> >>>>>> > >>>>>>>>>> >>>>>>>>> > >> --------------------------=
-----------------------------------------------> >> > >>>>>>>>> This SF.net=
 email is sponsored by DB2 Express> >>>>>>>>> Download DB2 Express C - the =
FREE version of DB2 express and> >>>>>>>>> > >>>>>> take> >>>>>> > >>>>>>>>=
> control of your XML. No limits. Just data. Click to get> >>>>>>>>> > >>>>=
 it now.> >>>> > >>>>>>>>> http://sourceforge.net/powerbar/db2/> >>>>>>>>> =
_______________________________________________> >>>>>>>>> opensipstack-dev=
el mailing list> >>>>>>>>> ope...@li...> >>>>>>=
>>>> >>>>>>>>> > >>>> https://lists.sourceforge.net/lists/listinfo/opensips=
tack-devel> >>>> > >>>>>>>>> >>>>>>>> > >> --------------------------------=
----------------------------------------> >> > >>>>>>>> Se uno de los prime=
ros en probar Windows Live Mail. Windows> >>>>>>>> > >>>>>> Live Mail.> >>>=
>>> > >> <http://ideas.live.com/programpage.aspx?versionId=3D5d21c51a-b161-=
4314-9b0e-4911fb2b2e6d>> >> > >>>>>>> >>>>>> > >> -------------------------=
-----------------------------------------------> >> > >>>>>> Se uno de los =
primeros en probar Windows Live Mail. Windows Live> >>>>>> Mail.> >>>>>>> >=
>>>>> > >> <http://ideas.live.com/programpage.aspx?versionId=3D5d21c51a-b16=
1-4314-9b0e-4911fb2b2e6d>> >> > >>>>>>> >>>>>>> >>>>>> > >> ---------------=
---------------------------------------------------------> >> > >>>>>> Se u=
no de los primeros en probar Windows Live Mail. Windows> >>>>>> > >>>> Live=
 Mail.> >>>> > >> <http://ideas.live.com/programpage.aspx?versionId=3D5d21c=
51a-b161-4314-9b0e-4911fb2b2e6d>> >> > >>>>>> >>>>> > >> ------------------=
-------------------------------------------------------> >> > >>>>> This SF=
.net email is sponsored by DB2 Express> >>>>> Download DB2 Express C - the =
FREE version of DB2 express and take> >>>>> control of your XML. No limits.=
 Just data. Click to get it now.> >>>>> http://sourceforge.net/powerbar/db2=
/> >>>>> _______________________________________________> >>>>> opensipstac=
k-devel mailing list> >>>>> ope...@li...> >>>>>=
 https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>>> > >=
>>>> >>>> > >> ------------------------------------------------------------=
------------> >> > >>>> Se uno de los primeros en probar Windows Live Mail.=
 Windows Live> >>>> Mail.> >>>>> >>>> > >> <http://ideas.live.com/programpa=
ge.aspx?versionId=3D5d21c51a-b161-4314-9b0e-4911fb2b2e6d>> >> > >>>>> >>>>>=
 >>>> > >> ----------------------------------------------------------------=
--------> >> > >>>> Comun=EDcate al instante con Windows Live Messenger Win=
dows Live> >>>> Messenger> >>>>> >>>> > >> <http://imagine-msn.com/messenge=
r/launch80/default.aspx?locale=3Des-ar&source=3Djoinmsncom/messenger> > >>>=
 >> > >>>> > >> -----------------------------------------------------------=
-------------> >> > >>>> > >> ---------------------------------------------=
----------------------------> >> > >>>> This SF.net email is sponsored by D=
B2 Express> >>>> Download DB2 Express C - the FREE version of DB2 express a=
nd take> >>>> control of your XML. No limits. Just data. Click to get it no=
w.> >>>> http://sourceforge.net/powerbar/db2/> >>>>> >>>> > >> ------------=
------------------------------------------------------------> >> > >>>> ___=
____________________________________________> >>>> opensipstack-devel maili=
ng list> >>>> ope...@li...> >>>> https://lists.=
sourceforge.net/lists/listinfo/opensipstack-devel> >>>>> >>>> > >>> --> >>>=
 Joegen E. Baclor> >>> CTO - Solegy LLC> >>>> >>> Email: joegen @ solegy.co=
m> >>> Main: +1 212 801 2504> >>> Fax: +1 347 438 3072> >>> Manila: +63 2 7=
47 3460> >>> Mobile: +63 918 411 9064> >>>> >>> 121 Varick St., Suite 201> =
>>> NY, NY 10013> >>>> >>> SOLEGY LLC> >>> http://www.solegy.com> >>> Solut=
ions to Fit Your Strategy> >>>> >>>> >>> > >> -----------------------------=
-------------------------------------------> >> Se uno de los primeros en p=
robar Windows Live Mail. Windows Live Mail. > >> <http://ideas.live.com/pro=
grampage.aspx?versionId=3D5d21c51a-b161-4314-9b0e-4911fb2b2e6d>> >> > >> >>=
 >> > ---------------------------------------------------------------------=
----> > This SF.net email is sponsored by DB2 Express> > Download DB2 Expre=
ss C - the FREE version of DB2 express and take> > control of your XML. No =
limits. Just data. Click to get it now.> > http://sourceforge.net/powerbar/=
db2/> > _______________________________________________> > opensipstack-dev=
el mailing list> > ope...@li...> > https://list=
s.sourceforge.net/lists/listinfo/opensipstack-devel> >> > > > > -----------=
--------------------------------------------------------------> This SF.net=
 email is sponsored by DB2 Express> Download DB2 Express C - the FREE versi=
on of DB2 express and take> control of your XML. No limits. Just data. Clic=
k to get it now.> http://sourceforge.net/powerbar/db2/> ___________________=
____________________________> opensipstack-devel mailing list> opensipstack=
-d...@li...> https://lists.sourceforge.net/lists/listinfo/=
opensipstack-devel
_________________________________________________________________
Descubre Live.com - tu mundo en l=EDnea reunido: noticias, deportes, el tie=
mpo, y mucho m=E1s.
http://www.live.com/getstarted=

[OpenSIPStack] SipUri Parse Error

[Gustavo C. <cur...@ho...>]

Joegen:When an uri like:<sip:782706@192.168.0.1:5060;transport=3Dudp;user=
=3Dphone>is parsed by SIPURI & SIPURI::operator=3D(const MimeHeader & heade=
r)the '>' is kept in the param user (user=3Dphone>). Then the result uri is=
:sip:782706@192.168.0.1:5060;transport=3Dudp;user=3Dphone>When this uri is =
parsed again by SIPURI & SIPURI::operator=3D(const MimeHeader & header),the=
 uri is detected like it has enclosure, but the uri is not extracted.I made=
 a few changes to correct the second error:PINDEX begin =3D value.Find( '<'=
 ) + 1; --> PINDEX begin =3D value.Find( '<' ); (to be compared with P_MAX_=
INDEX)PINDEX end =3D value.Find( '>' );BOOL hasEnclosure =3D begin !=3D P_M=
AX_INDEX && end !=3D P_MAX_INDEX; PString uri;if( hasEnclosure )=20
{=20
 PINDEX rightOffSet =3D value.GetLength() - end;=20
 PINDEX length =3D value.GetLength() - begin - rightOffSet; --> PINDEX leng=
th =3D value.GetLength() - begin - 1 - rightOffSet;=20
 uri =3D value.Mid( begin, length ).Trim(); --> uri =3D value.Mid( begin + =
1, length ).Trim();=20
} And to correct the first error: PStringArray t1, t2;=20
PString tBuff =3D value;PINDEX atIndex =3D value.FindLast( '@' ); if( atInd=
ex !=3D P_MAX_INDEX ) tBuff =3D value.Mid( atIndex + 1 );|VPStringArray t1,=
 t2;=20
PString tBuff =3D value;PINDEX atIndex =3D value.FindLast( '@' );if( atInde=
x !=3D P_MAX_INDEX )=20
{
 if( !hasEnclosure )=20
 {=20
  tBuff =3D value.Mid( atIndex + 1 );=20
 } else {=20
  atIndex =3D uri.FindLast( '@' );=20
  tBuff =3D uri.Mid( atIndex + 1 );=20
 }=20
}
Please check if these changes are ok.  Regards,
Gustavo
_________________________________________________________________
Descubre Live.com - tu propia p=E1gina de inicio, personalizada para ver r=
=E1pidamente todo lo que te interesa en un mismo sitio.
http://www.live.com/getstarted=

[OpenSIPStack] SipUri Parse Error

[Gustavo C. <cur...@ho...>]

Joegen:
=20
When an uri like:
=20
<sip:782706@192.168.0.1:5060;transport=3Dudp;user=3Dphone>
=20
is parsed by SIPURI & SIPURI::operator=3D(const MimeHeader & header)
=20
the '>' is kept in the param user (user=3Dphone>).=20
=20
Then the result uri is:
=20
sip:782706@192.168.0.1:5060;transport=3Dudp;user=3Dphone>
=20
When this uri is parsed again by SIPURI & SIPURI::operator=3D(const MimeHea=
der & header),
=20
the uri is detected like it has enclosure, but the uri is not extracted.
=20
I made a few changes to correct the second error:
=20
 PINDEX begin =3D value.Find( '<' ) + 1; --> PINDEX begin =3D value.Find( '=
<' ); (to be compared with P_MAX_INDEX)
 PINDEX end =3D value.Find( '>' );
 BOOL hasEnclosure =3D begin !=3D P_MAX_INDEX && end !=3D P_MAX_INDEX; PStr=
ing uri;
=20
 if( hasEnclosure ) {   PINDEX rightOffSet =3D value.GetLength() - end;   P=
INDEX length =3D value.GetLength() - begin - rightOffSet; --> PINDEX length=
 =3D value.GetLength() - begin - 1 - rightOffSet;   uri =3D value.Mid( begi=
n, length ).Trim(); --> uri =3D value.Mid( begin + 1, length ).Trim(); }
=20
And to correct the first error:
=20
 PStringArray t1, t2; PString tBuff =3D value;
 PINDEX atIndex =3D value.FindLast( '@' );
=20
 if( atIndex !=3D P_MAX_INDEX )
  tBuff =3D value.Mid( atIndex + 1 );
=20
                 |
                 V
 PStringArray t1, t2; PString tBuff =3D value;
 PINDEX atIndex =3D value.FindLast( '@' );
=20
 if( atIndex !=3D P_MAX_INDEX ) {  if( !hasEnclosure )  {   tBuff =3D value=
.Mid( atIndex + 1 );  } else {   atIndex =3D uri.FindLast( '@' );   tBuff =
=3D uri.Mid( atIndex + 1 );  } }
Please check if these changes are ok.
=20
Regards,Gustavo
_________________________________________________________________
Descubre Live.com - tu mundo en l=EDnea reunido: noticias, deportes, el tie=
mpo, y mucho m=E1s.
http://www.live.com/getstarted=

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Ehh I celebrate to fast :)

When I change ringing sound then also sound (tone) that I hear in ATLSoftphone changes. I am interested to hear different sound when somebody call to me and hear different when i call to remote subscriber. Is it possible at all?

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Hey!

OK everytihgn works great!

Thank you!

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[Ilian J. C. P. <ip...@so...>]

Hi,

tomach wrote:
> Sorry, when I unregister and downlad whole cvs again not just update everythign works fine.
>
> Well done! 
>
> Now I just have to find out how to solve this ring problem cos I did as you adviced and (with audio file and change lines of code but it didnt help) :(. Are you sure that  you are albe to chage the sound of ring?
>   
Yep. Just tried it a few moments ago. Just make sure that your .wav is 
16 bit 8 kHz Mono.

Also, if this is how you coded it exactly:

PFilePath pf("C:\ringin.wav");
m_RingBackToneFile = new MS::VoiceFile();
m_RingBackToneFile->Open(pf, MS::VoiceFile::Wav_8khz_Mono,"RING_XXX" );

You may have a problem with "C:\ringin.wav", you're missing one '\'. :)

- Ilian
> By the way I also checked how applicaotin looks in memory profiles and it looks quite good:)
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Sorry, when I unregister and downlad whole cvs again not just update everythign works fine.

Well done! 

Now I just have to find out how to solve this ring problem cos I did as you adviced and (with audio file and change lines of code but it didnt help) :(. Are you sure that  you are albe to chage the sound of ring?

By the way I also checked how applicaotin looks in memory profiles and it looks quite good:)

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[Ilian J. C. P. <ip...@so...>]

Hi Tom,

Everything works fine in my end.

Let's try to localize the problem. Did you update OpalOSSConnection.cxx 
only? Or did you update everything in OpenSIPStack?

Regards,
Ilian

tomach wrote:
> Hi Ilian,
>
> hmmm this time it fails totally when someobdy call to atlsip, aplication just quits :( with error and devbugger can not find it unfortunatly...whats more when I call somebody I can not here voice in my speakers....only the remote side can here me. So I guess its better to come back to previous version and update sometihng else....
>
> I am waiting for news then...
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[tomach <to...@dg...>]

Hi Ilian,

hmmm this time it fails totally when someobdy call to atlsip, aplication just quits :( with error and devbugger can not find it unfortunatly...whats more when I call somebody I can not here voice in my speakers....only the remote side can here me. So I guess its better to come back to previous version and update sometihng else....

I am waiting for news then...