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From: Andre S. <eds...@ya...> - 2007-07-17 01:34:34
|
Will my curent ATLSIP work with this new version? "Joegen E. Baclor" <joe...@gm...> wrote: Hi Everyone, First, I apologize for the volume of unanswered e-mails in the mailing list. I have been busy with the July 15th Russell release. Version 1.1.4 of OpenSBC is now available in CVS. This version is going to be tagged as the "Russell" release within the next few days as soon as our internal Q.A. has given the go signal. For the meantime, the CVS is again open for minor bug fixes. Feel free to test the new version and notify me rightaway if you encounter any trouble. Among the things you would notice in this release is the use of a new STL based String class which is 5 times faster than the PWLIB PString class. We have seen that we have regained 16% of CPU load simply because of this change. You will see more of the container classes being migrated to their STL couter-parts in the future. There is also a new section in the HTTP admin that displays numerous resource counters such as volume of transactions, calls, connections, registrations etc. The download section in opensipstack will be updated within the week. We also plan to reactivate the source forge download pages. User documentations will be published together with the downloads. Cheers! Joegen ------------------------------------------------------------------------- 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 --------------------------------- We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-16 12:12:28
|
Hi Tom,
All logging (previously opal.log and sip.log) can now be centralized to
the Logger class (courtesy of Joegen). All you have to do is replace the
following lines in SoftPhone.cxx:
Logger::SetDefaultLevel( 5 );
Logger::SetDefaultLogFile( "sip.log" );
If I wanted to redirect ALL logs to the sip.log file, I could replace
the lines with:
Logger::SetDefaultLevel( 5 );
static unsigned int options = // These are the options set in
SetDefaultLogFile
LogDateAndTime |
LogThread |
LogAppendToFile;
Logger::SetDefaultLogStream( new LoggingFileStream( "sip.log",
options ) ); // Redirects all logs to sip.log
If you want to redirect the logs to some place else, just subclass
LoggingDestination and pass a pointer to that object in
SetDefaultLogStream. All classes you need for extending the logging
behavior is found in Logger.cxx. You can also find built-in
LoggingDestination subclassess in the same file which you may find useful.
Also, please wait for Joegen's announcement of the OpenSIPStack stable
release. There are many modifications. So you will have to fully
checkout the modules then.
Regards,
Ilian
tomach wrote:
> thanks for the answers
>
> Ok, then please try to make a solution because on the begging of next week i give it to test and would be good to have logs during this test....that to know what goes wrong in case...
>
> -------------------------------------------------------------------------
> 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
>
>
>
|
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-16 12:03:48
|
Hi Tom,
All logging (previously opal.log and sip.log) can now be centralized to
the Logger class (courtesy of Joegen). All you have to do is replace the
following lines in SoftPhone.cxx:
Logger::SetDefaultLevel( 5 );
Logger::SetDefaultLogFile( "sip.log" );
If I wanted to redirect ALL logs to the sip.log file, I could replace
the lines with:
Logger::SetDefaultLevel( 5 );
static unsigned int options = // These are the options set in
SetDefaultLogFile
LogDateAndTime |
LogThread |
LogAppendToFile;
Logger::SetDefaultLogStream( new LoggingFileStream( "sip.log",
options ) ); // Redirects all logs to sip.log
If you want to redirect the logs to some place else, just subclass
LoggingDestination and pass a pointer to that object in
SetDefaultLogStream. All classes you need for extending the logging
behavior is found in Logger.cxx. You can also find built-in
LoggingDestination subclassess in the same file which you may find useful.
Also, please wait for Joegen's announcement of the OpenSIPStack stable
release. There are many modifications. So you will have to fully
checkout the modules then.
Regards,
Ilian
tomach wrote:
> thanks for the answers
>
> Ok, then please try to make a solution because on the begging of next week i give it to test and would be good to have logs during this test....that to know what goes wrong in case...
>
> -------------------------------------------------------------------------
> 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
>
>
>
|
|
From: Joegen E. B. <joe...@gm...> - 2007-07-16 11:53:32
|
Hi Everyone, First, I apologize for the volume of unanswered e-mails in the mailing list. I have been busy with the July 15th Russell release. Version 1.1.4 of OpenSBC is now available in CVS. This version is going to be tagged as the "Russell" release within the next few days as soon as our internal Q.A. has given the go signal. For the meantime, the CVS is again open for minor bug fixes. Feel free to test the new version and notify me rightaway if you encounter any trouble. Among the things you would notice in this release is the use of a new STL based String class which is 5 times faster than the PWLIB PString class. We have seen that we have regained 16% of CPU load simply because of this change. You will see more of the container classes being migrated to their STL couter-parts in the future. There is also a new section in the HTTP admin that displays numerous resource counters such as volume of transactions, calls, connections, registrations etc. The download section in opensipstack will be updated within the week. We also plan to reactivate the source forge download pages. User documentations will be published together with the downloads. Cheers! Joegen |
|
From: Arif V. K. <msg...@gm...> - 2007-07-13 18:50:03
|
Hello, I am wondering after the compilation of Opensipstack solution what i can do further, your documentation didn't specify how to run this solution(Opensipstack) after build and compilation. I need to know how to run the opensipstack and I want to see if you have any running application based on opensipstack or developed with opensipstack. Again need to know how to run the solution OpenSIPstack after compilation and build and also want to look at any application based on OpenSIPStack. Arif Research Assistant Electrical and Computer Engineering Concordia University Montreal, Canada On 7/8/07, Joegen E. Baclor <joe...@gm...> wrote: > */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 > > > > ------------------------------------------------------------------------- > > 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 > |
|
From: tomach <to...@dg...> - 2007-07-13 12:51:18
|
thanks for the answers Ok, then please try to make a solution because on the begging of next week i give it to test and would be good to have logs during this test....that to know what goes wrong in case... |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-13 11:52:30
|
Hi. tomach wrote: > Why I need logs? > > I need them cos when let say 20 people would use ATLSIP at the same time and if one will fall down I would like to have some proofe why somethign went wrong. I belive that You have really a lot information in those logs...and I am sure that even if I would like to ask you for help you would ask me for logs...and I can not let logs to files like it was used to done...Anyway I need those logs in case of failure. > Yes I understand why you need them. Currently, there is no provision for redirecting the info from opal.log and sip.log. I'll try to think of a way. Please wait for updates. > Does those logs from Event_ReadPacketLog and Event_WritePacketLog would be enough for You for example to anlazyze? > You will only get SIP messages here. So this may not be sufficient for your case. For OSSPhone, you should receive these events in ATLSIP_OnLogSIPMessage(). Regards, Ilian > Else where can I find those events? Because they are not provided in ATLSIP activex....Is it possible to put them there? > > ------------------------------------------------------------------------- > 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 > > > |
|
From: tomach <to...@dg...> - 2007-07-13 10:43:40
|
Why I need logs? I need them cos when let say 20 people would use ATLSIP at the same time and if one will fall down I would like to have some proofe why somethign went wrong. I belive that You have really a lot information in those logs...and I am sure that even if I would like to ask you for help you would ask me for logs...and I can not let logs to files like it was used to done...Anyway I need those logs in case of failure. Does those logs from Event_ReadPacketLog and Event_WritePacketLog would be enough for You for example to anlazyze? Else where can I find those events? Because they are not provided in ATLSIP activex....Is it possible to put them there? |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-13 09:41:00
|
Hi, tomach wrote: > Hello Ilian!! > > First of all I decided to test ATL with many people at the same time. Because we belive it is really good application. > But right now I have big request to You. I would like to catch all your logs to our system. That in case of failure I analyze logs... > I have my own log system. Is it possible that ATL and opensipstac would send all logs not ot file but to the event that i woudl catch in my softphone ?? Is it easy task and would work properly? What woudl be the most changes? > > In case its too complicated, how to diseable logs totally? Because I notice that files grow really fast and its dangerous for my system. > Disabling the logs will be as simple as removing the following lines from SoftPhone.cxx: PTrace::Initialise( 3, "opal.log" ); ===================================== Logger::SetDefaultLevel( 5 ); Logger::SetDefaultLogFile( "sip.log" ); When disabling these, you can still get the SIP messages from Event_ReadPacketLog and Event_WritePacketLog but there is no provision yet for getting the info from sip.log and opal.log. Do you need those info too? Regards, Ilian > Please consider my questions its very important for me to start really tests, that could be really valuable for You. > > Best Regards, > Tomek > > ------------------------------------------------------------------------- > 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 > > > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-13 09:29:26
|
OK. Just please wait until after the OpenSIPStack stable release (July 15 I think). I'll check-in the mods by then. Thanks! Regards, Ilian Andre Silo wrote: > Ilian, > > Problem solved! I just set the SetSoundChannelBufferDepth() from 5 to 10. Could I request to put out this method from ATLSIP > > "Ilian Jeri C. Pinzon" <ip...@so...> wrote: > Hello Andre, > > You forgot to attach the RTP log. Also, have you tried changing the > silence detection method? Vista has may changes in its sound API. This > might have messed up the adaptive silence detection algo. > > Regards, > Ilian > > Andre Silo wrote: > >> Hello, >> >> I tried using compiling opensipstack on Vista Home Premium but I encounter to much chopiness on its voice. Could you determine why? Attached is the RTP of this call >> >> >> --------------------------------- >> Bored stiff? Loosen up... >> Download and play hundreds of games for free on Yahoo! Games. >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> 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 >> >> ------------------------------------------------------------------------ >> >> No virus found in this incoming message. >> Checked by AVG Free Edition. >> Version: 7.5.476 / Virus Database: 269.10.4/897 - Release Date: 7/11/2007 9:57 PM >> >> > > > ------------------------------------------------------------------------- > 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 > > > > --------------------------------- > Need a vacation? Get great deals to amazing places on Yahoo! Travel. > ------------------------------------------------------------------------- > 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 > > > |
|
From: Andre S. <eds...@ya...> - 2007-07-13 09:07:15
|
Ilian, Problem solved! I just set the SetSoundChannelBufferDepth() from 5 to 10. Could I request to put out this method from ATLSIP "Ilian Jeri C. Pinzon" <ip...@so...> wrote: Hello Andre, You forgot to attach the RTP log. Also, have you tried changing the silence detection method? Vista has may changes in its sound API. This might have messed up the adaptive silence detection algo. Regards, Ilian Andre Silo wrote: > Hello, > > I tried using compiling opensipstack on Vista Home Premium but I encounter to much chopiness on its voice. Could you determine why? Attached is the RTP of this call > > > --------------------------------- > Bored stiff? Loosen up... > Download and play hundreds of games for free on Yahoo! Games. > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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 > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.476 / Virus Database: 269.10.4/897 - Release Date: 7/11/2007 9:57 PM > ------------------------------------------------------------------------- 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 --------------------------------- Need a vacation? Get great deals to amazing places on Yahoo! Travel. |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-13 09:03:26
|
Hello Andre, You forgot to attach the RTP log. Also, have you tried changing the silence detection method? Vista has may changes in its sound API. This might have messed up the adaptive silence detection algo. Regards, Ilian Andre Silo wrote: > Hello, > > I tried using compiling opensipstack on Vista Home Premium but I encounter to much chopiness on its voice. Could you determine why? Attached is the RTP of this call > > > --------------------------------- > Bored stiff? Loosen up... > Download and play hundreds of games for free on Yahoo! Games. > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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 > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.476 / Virus Database: 269.10.4/897 - Release Date: 7/11/2007 9:57 PM > |
|
From: tomach <to...@dg...> - 2007-07-13 08:44:15
|
Hello Ilian!! First of all I decided to test ATL with many people at the same time. Because we belive it is really good application. But right now I have big request to You. I would like to catch all your logs to our system. That in case of failure I analyze logs... I have my own log system. Is it possible that ATL and opensipstac would send all logs not ot file but to the event that i woudl catch in my softphone ?? Is it easy task and would work properly? What woudl be the most changes? In case its too complicated, how to diseable logs totally? Because I notice that files grow really fast and its dangerous for my system. Please consider my questions its very important for me to start really tests, that could be really valuable for You. Best Regards, Tomek |
|
From: Gowri K. <Gow...@in...> - 2007-07-13 08:41:13
|
Hi All, =20 I am new to SIP. I am trying to build an user agent application with opensipstack. Please let me know on how to use the library. Any pointers to documents would be of great help. =20 Thanks, Gowri **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended= solely for the use of the addressee(s). If you are not the intended= recipient, please notify the sender by e-mail and delete the original= message. Further, you are not to copy, disclose, or distribute this e-mail= or its contents to any other person and any such actions are unlawful.= This e-mail may contain viruses. Infosys has taken every reasonable= precaution to minimize this risk, but is not liable for any damage you may= sustain as a result of any virus in this e-mail. You should carry out your= own virus checks before opening the e-mail or attachment. Infosys reserves= the right to monitor and review the content of all messages sent to or= from this e-mail address. Messages sent to or from this e-mail address may= be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS*** |
|
From: Joegen E. B. <joe...@gm...> - 2007-07-12 09:17:44
|
MPL is the default license for opensipstack. GPL/LGPL are alternative licenses. The reason we chose to go tipple license is because of the inherent incompatibility between MPL and GPL. MPL allows for applications using the library to be closed source as long as there is no changes made to the library, while GPL is viral and requires all derived work to be open-source. The motivation behind the MPL is to allow application developers to use proprietary libraries such as codecs that cannot be published in open-source because of patent or copyright issues or simply because the library reveals trade secrets of the company using the library. For those whose applications are already under the GPL/LGPL they could simply choose the alternate license without having to get our permission because the license readily allows it. John Rexie Mercado wrote: > Hi! > > Why is that OpenSipStack under a triple license? > Is GPL not sufficient? > And why the licenses you have chosen, why not other open source licenses? > > Thanks! > |
|
From: Andre S. <eds...@ya...> - 2007-07-12 08:59:30
|
Hello, I tried using compiling opensipstack on Vista Home Premium but I encounter to much chopiness on its voice. Could you determine why? Attached is the RTP of this call --------------------------------- Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-12 04:47:08
|
Hi, The road map on the web is for OSBC. Those features are not related to OSSPhone. Regards, Ilian Giuseppe Battaglia wrote: > Hi, > I'm interested in IVR and VXML support, both present in the road map > published on the web, but not present in your last list. > Have you removed these features or are not related to OSSPhone? > Thank you very much, > Giuseppe Battaglia > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of Ilian > Jeri C. Pinzon > Sent: martedì 10 luglio 2007 11.28 > To: ope...@li... > Subject: [OpenSIPStack] OSSPhone road map > > Hello Everyone, > > I'm planning to include the following features in OSSPhone. If you wish > some feature to be included or if you have any suggestions, please don't > hesitate to ask. Maybe I could make a room for it somewhere. :) > > Thanks! > > ============================================== > > OSSPhone road map > > 1. Call transfer and call hold (July 27) > 2. USB phone support (August) > 3. Mac support (3Q) > 4. Multiple call lines (3Q) > 5. IM support (3Q) > 6. Playing of audio files over the wire (4Q) > 7. Video support (4Q) > > Regards, > Ilian > > > ------------------------------------------------------------------------- > 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 > > > |
|
From: Gustavo C. <cur...@ho...> - 2007-07-11 17:47:19
|
Hi Joegen:
=20
I made changes in the code to avoid that the invite continue being sent whe=
n sesion is being cancelled.
When a cancel is received, after cancelling the Invite Server Transaction, =
I terminate the Invite Client Transaction:
=20
BOOL ProxySession::ValidateRequest(SIPMessage & request)
{
....
}else if( request.IsCancel() ) { m_SessionManager.GetUserAgent().GetS=
tack().CancelInviteServerTransaction(m_Request); m_SessionManager.GetUse=
rAgent().GetStack().TerminateInviteClientTransaction(m_CurrentUACRequest); =
}
}
=20
BOOL SIPTransactionManager::TerminateInviteClientTransaction(const SIPMessa=
ge & invite){ if( m_IsTerminating ) return FALSE;
=20
PAssert( invite.IsInvite(), PLogicError );
=20
PWaitAndSignal lock( m_TransactionPoolMutex );
=20
TransactionId id =3D invite.GetTransactionId();
=20
if( id.AsString().IsEmpty() ) { LOG_IF_DEBUG( LogInfo(), "TerminateIn=
viteClientTransaction() - Unable to match transaction because transaction-i=
d is empty" ); return FALSE; }
=20
id.SetStateMachine( "ICT" );
=20
SIPTransaction * transaction =3D m_TransactionPool.GetAt( id.AsString() )=
; if( transaction =3D=3D NULL ) { return FALSE; } transaction->Ter=
minate();
=20
return TRUE;}
=20
And in ProxySession::OnDNSFailOver(), the m_CurrentUACRequest is actualized=
:
=20
void ProxySession::OnDNSFailOver(SIPDNSFailOver & failOverEvent){ SIPMessag=
e msg =3D failOverEvent.GetMessage();
=20
if( msg.IsInvite() ) { m_CurrentUACRequest =3D msg.AsString(); SIPURI ta=
rgetURI; msg.GetRequestURI(targetURI); SetDialogPeerAddress( targetURI.As=
String() ); }}
First, I tried:m_CurrentUACRequest =3D msg;But the m_CurrentUACRequest was =
not actualized, because the=20
m_InternalId was the same for both messages and:
=20
SIPMessage & SIPMessage::operator=3D(const SIPMessage & msg)
{
if( msg.m_InternalId =3D=3D m_InternalId )
return *this;
...
}
=20
Then i used the AsString():
=20
m_CurrentUACRequest =3D msg.AsString();
=20
I know this is not the best solution, then if you could suggest a better on=
e please let me know. Maybe in=20
SIPTransaction::DoDNSFailover(), changing :
=20
SIPMessage request =3D m_OpeningRequest;
=20
I attach the sip flows before and after the changes in the code, the second=
sip flow is more normal.
=20
Thanks for your help.
=20
Gustavo
=20
=20
> Date: Wed, 4 Jul 2007 15:06:09 +0800> To: cur...@gm...> From=
: joe...@gm...> CC: ope...@li...> Su=
bject: Re: [OpenSIPStack] FW: FW: OpenSBC as Forking Proxy> > Gustavo Curet=
ti wrote:> > Hi Joegen:> > > > I don't have a call sesion, i have a proxy s=
ession. The cancel is sent > > to the m_DialogPeerAddress for a proxy sessi=
on. I have added the > > following code:> > > > void ProxySession::OnDNSFai=
lOver(> > SIPDNSFailOver & failOverEvent> > )> > {> > SIPMessage msg =3D fa=
ilOverEvent.GetMessage();> > if( msg.IsInvite() )> > {> > SIPURI targetURI;=
> > msg.GetRequestURI(targetURI);> > SetDialogPeerAddress( targetURI.AsStri=
ng() );> > }> > }> > I have patched CVS accordingly.> > > > > > Then the ca=
ncel is sent to the actual target of the ICT transaction, > > but if the ac=
tual target is not answering, the ICT continue trying > > with the actual t=
arget and with the followings. The ICT transaction > > must stop trying. Co=
uld you give some directions, please?> > I think this is a compliant behavi=
or. Don't you want the UA to know > that the UAS is actually down? I have c=
hecked 3261 again and paragraph > indicates that a proxy MAY send a respons=
e to the CANCEL and generate a > new transaction towards the UAS. This is n=
ot a MUST and is left to the > implementors to decide on. I am open to disc=
uss this further if you > think it is a better approach to handle extra sta=
te 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 tha=
t way.> > 9.2 Server Behavior> > The CANCEL method requests that the TU at =
the server side cancel a> pending transaction. The TU determines the transa=
ction to be> cancelled by taking the CANCEL request, and then assuming that=
the> request method is anything but CANCEL or ACK and applying the> transa=
ction matching procedures of Section 17.2.3. The matching> transaction is t=
he one to be cancelled.> > The processing of a CANCEL request at a server d=
epends on the type of> server. A stateless proxy will forward it, a statefu=
l 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 CAN=
CEL.> > > > > >> > 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: OpenSB=
C as Forking Proxy> > >> > > Hi Gustavo,> > >> > > I have committed a fix o=
f 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. Lem=
me dig further. I'll let you know once a fix is> > > > available.> > > >> >=
> >> > > > Gustavo Curetti wrote:> > > >> > > >> Joegen:> > > >>> > > >> T=
he fix works fine. But i have the following problem: When i hang up> > > >>=
and send a Cancel to OpenSBC, the OpenSBC respond with> > > >> Code487_Req=
uestCancelled but the Cancel is not been send to the > > actual> > > >> tar=
get, the Cancel is just routed again.> > > >>> > > >> Thanks for your help.=
> > > >>> > > >> Gustavo> > > >>> > > >> > > ------------------------------=
------------------------------------------> > > >>> > > >>> > > >>> Date: W=
ed, 13 Jun 2007 22:15:14 +0800> > > >>> To: cur...@gm...; > > =
ope...@li...> > > >>> Subject: Re: [OpenSIPStac=
k] 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_FailOv=
erAttempts> > > >>>> is increased from 0 to 1.> > > >>>>> > > >>>> Then the=
via is changed (+ "-") and with this change when> > > >>>> FindTransaction=
AndAddEvent() is called a new transaction is > > created.> > > >>>> In SIPT=
ransaction::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: OpenSB=
C 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: curetti.gustavo@gm=
ail.com;> > > >>>>>> > > >>>> 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 h=
ave committed this a few days ago to> > > >>>>> demonstrate fail-over by fo=
rking using DNS/SRV records. New> > > >>>>> transactions are created by cal=
ling> > > >>>>>> > > >>>> 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 cl=
ient transaction when> > > >>>>>>> > > >>>> the> > > >>>>> > > >>>>>> first=
invite fail in the FSM layer. Could you give some> > > >>>>>>> > > >>>> di=
rections,> > > >>>>> > > >>>>>> please?> > > >>>>>>> > > >>>>>> Thanks for =
your help.> > > >>>>>>> > > >>>>>> Gustavo.> > > >>>>>>> > > >>>>>>> > > >>=
>>>>> > > >>>>>>> > > >>>>>>> > > >> > > ----------------------------------=
--------------------------------------> > > >>> > > >>>>>> From: curetti_gu=
st...@ho...> > > >>>>>> To: jb...@so...> > > >>>>>> Sub=
ject: 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> > > >>>>>> Gust=
avo> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >> > > -----------------------=
-------------------------------------------------> > > >>> > > >>>>>>> Date=
: Thu, 10 May 2007 13:28:29 +0800> > > >>>>>>> From: jbaclor@solegysystems.=
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 s=
pawning an> > > >>>>>>>> > > >>>>>> outbound> > > >>>>>>> > > >>>>>>> invit=
e. There should be a clean mechanism to clone> > > >>>>>>>> > > >>>> transa=
ctions 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 co=
de for timer B> > > >>>>>>>>> > > >>>>>> expiration just> > > >>>>>>> > > >=
>>>>>>> for do some tests:> > > >>>>>>>>> > > >>>>>>>> void ProxySessionMan=
ager::OnTimerExpire(> > > >>>>>>>> SIPTimerExpire & timerEvent,> > > >>>>>>=
>> SIPSession * session> > > >>>>>>>> )> > > >>>>>>>> {> > > >>>>>>>> if( s=
ession !=3D NULL )> > > >>>>>>>> {> > > >>>>>>>> LOG_IF_DEBUG( LogWarning()=
, "*** TIMER EXPIRATION *** for SIP> > > >>>>>>>> Session " << session->Get=
SessionId() );> > > >>>>>>>> if( timerEvent.GetTimer() =3D=3D> > > >>>>>>>>=
> > > >>>> SIPTransactions::SIPTimerEvent::B)> > > >>>>> > > >>>>>>>> {> > =
> >>>>>>>> SIPMessage msg =3D ((ProxySession> > > >>>>>>>>> > > >>>> *)sess=
ion)->GetOriginalInvite();> > > >>>>> > > >>>>>>>> session->EnqueueSessionE=
vent( new SIPSessionEvent(> > > >>>>>>>>> > > >>>> *session, 1,> > > >>>>> =
> > >>>>>>>> msg ) );> > > >>>>>>>> }> > > >>>>>>>> session->OnTimerExpire(=
timerEvent );> > > >>>>>>>> }> > > >>>>>>>> }> > > >>>>>>>>> > > >>>>>>>> =
and> > > >>>>>>>>> > > >>>>>>>> void ProxySession::OnTimerExpire(> > > >>>>=
>>>> SIPTimerExpire & timerEvent> > > >>>>>>>> )> > > >>>>>>>> {> > > >>>>>=
>>> GCREF( "SIPSession::OnTimerExpire" );> > > >>>>>>>> if( timerEvent.GetT=
imer() =3D=3D> > > >>>>>>>>> > > >>>> SIPTransactions::SIPTimerEvent::B ||>=
> > >>>>> > > >>>>>>>> timerEvent.GetTimer() =3D=3D SIPTransactions::SIPTi=
merEvent::F )> > > >>>>>>>> {> > > >>>>>>>> ///this is an ICT timeout> > > =
>>>>>>>> SIPMessage timeout;> > > >>>>>>>> GetCurrentUASRequest().CreateRes=
ponse( timeout,> > > >>>>>>>> SIPMessage::Code480_TemporarilyNotAvailable )=
;> > > >>>>>>>> SendRequest( timeout );> > > >>>>>>>> }> > > >>>>>>>>> > > =
>>>>>>>> //Destroy();> > > >>>>>>>> }> > > >>>>>>>>> > > >>>>>>>> With thes=
e changes and a relay route:> > > >>>>>>>>> > > >>>>>>>> [sip:*@192.168.0.2=
07:*] 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 t=
o 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 n=
ew ICT for trying the next> > > >>>>>>>> destination but canceling throw th=
e first ICT at the same> > > >>>>>>>>> > > >>>> time.> > > >>>>> > > >>>>>>=
>> Thanks for your help.> > > >>>>>>>>> > > >>>>>>>> Gustavo> > > >>>>>>>>>=
> > >>>>>>>>> > > >>>>>>>>> > > >> > > -----------------------------------=
-------------------------------------> > > >>> > > >>>>>>>>> Date: Sat, 5 M=
ay 2007 13:50:44 +0800> > > >>>>>>>>> From: jb...@so...> > > =
>>>>>>>>> To: cur...@gm...;> > > >>>>>>>>>> > > >>>>>> opensip=
sta...@li...> > > >>>>>>> > > >>>>>>>>> Subject: Re: [=
OpenSIPStack] OpenSBC as Forking Proxy> > > >>>>>>>>>> > > >>>>>>>>> Gustav=
o,> > > >>>>>>>>>> > > >>>>>>>>> Forking is not supported yet in OpenSBC.> =
> > >>>>>>>>>> > > >>>>>>>>> Joegen> > > >>>>>>>>>> > > >>>>>>>>> Gustavo C=
uretti wrote:> > > >>>>>>>>>> > > >>>>>>>>>> Hi Joegen:> > > >>>>>>>>>>> > =
> >>>>>>>>>> I want to use the OpenSBC as a Forking Proxy. I want> > > >>>>=
>>>>>>> > > >>>> that the> > > >>>>> > > >>>>>>>>>> OpenSBC try the differe=
nts 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> > > >>>>>>>>> > > >=
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>>>>>>>> > > >>>>>> todo en el> > > >>>>>>> > > >>>>>>>>>> mismo sitio. <ht=
tp://www.live.com/getstarted>> > > >>>>>>>>>>> > > >>>>>>>>>>> > > >> > > -=
-----------------------------------------------------------------------> > =
> >>> > > >>>>>>>>>>> > > >> > > ------------------------------------------=
-------------------------------> > > >>> > > >>>>>>>>>> This SF.net email i=
s sponsored by DB2 Express> > > >>>>>>>>>> Download DB2 Express C - the FRE=
E version of DB2 express> > > >>>>>>>>>>> > > >>>>>> and take> > > >>>>>>> =
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rge.net/powerbar/db2/> > > >>>>>>>>>>> > > >>>>>>>>>>> > > >> > > ---------=
---------------------------------------------------------------> > > >>> > =
> >>>>>>>>>> _______________________________________________> > > >>>>>>>>>=
> opensipstack-devel mailing list> > > >>>>>>>>>> opensipstack-devel@lists.=
sourceforge.net> > > >>>>>>>>>>> > > >>>>>>>>>>> > > >>>>>> https://lists.s=
ourceforge.net/lists/listinfo/opensipstack-devel> > > >>>>>>> > > >>>>>>>>>=
> > > >>>>>>>>>> > > >> > > -----------------------------------------------=
--------------------------> > > >>> > > >>>>>>>>> This SF.net email is spon=
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r/db2/> > > >>>>>>>>> _______________________________________________> > > =
>>>>>>>>> opensipstack-devel mailing list> > > >>>>>>>>> opensipstack-devel=
@lists.sourceforge.net> > > >>>>>>>>>> > > >>>>>>>>>> > > >>>> https://list=
s.sourceforge.net/lists/listinfo/opensipstack-devel> > > >>>>> > > >>>>>>>>=
> > > >>>>>>>>> > > >> > > ------------------------------------------------=
------------------------> > > >>> > > >>>>>>>> Se uno de los primeros en pr=
obar Windows Live Mail. Windows> > > >>>>>>>>> > > >>>>>> Live Mail.> > > >=
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> ------------------------------------------------------------------------>=
> > >>> > > >>>>>> Se uno de los primeros en probar Windows Live Mail. Win=
dows Live> > > >>>>>> Mail.> > > >>>>>>> > > >>>>>>> > > >> > > <http://ide=
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6d>> > > >>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >> > > ---------------=
---------------------------------------------------------> > > >>> > > >>>>=
>> Se uno de los primeros en probar Windows Live Mail. Windows> > > >>>>>>>=
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>> > > >>>>>> > > >> > > --------------------------------------------------=
-----------------------> > > >>> > > >>>>> This SF.net email is sponsored b=
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_______________________________________________> > > >>>>> opensipstack-dev=
el 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=3D5d21c51a=
-b161-4314-9b0e-4911fb2b2e6d>> > > >>> > > >>>>> > > >>>>> > > >>>>> > > >>=
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com/messenger> > >> > > >>> > > >>> > > >>>>> > > >> > > ------------------=
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> >> > > -----------------------------------------------------------------=
--------> > > >>> > > >>>> This SF.net email is sponsored by DB2 Express> >=
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> > >>>> control of your XML. No limits. Just data. Click to get it now.> >=
> >>>> http://sourceforge.net/powerbar/db2/> > > >>>>> > > >>>>> > > >> > =
> ------------------------------------------------------------------------>=
> > >>> > > >>>> _______________________________________________> > > >>>>=
opensipstack-devel mailing list> > > >>>> ope...@li...urcef=
orge.net> > > >>>> https://lists.sourceforge.net/lists/listinfo/opensipstac=
k-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 =
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> >>>> > > >> > > --------------------------------------------------------=
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. Windows Live > > Mail.> > > >> > > <http://ideas.live.com/programpage.asp=
x?versionId=3D5d21c51a-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> > > > c=
ontrol of your XML. No limits. Just data. Click to get it now.> > > > http:=
//sourceforge.net/powerbar/db2/> > > > ____________________________________=
___________> > > > opensipstack-devel mailing list> > > > opensipstack-deve=
l...@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. Cli=
ck to get it now.> > > http://sourceforge.net/powerbar/db2/> > > __________=
_____________________________________> > > opensipstack-devel mailing list>=
> > ope...@li...> > > https://lists.sourceforg=
e.net/lists/listinfo/opensipstack-devel> >> >> > --------------------------=
----------------------------------------------> > Env=EDa mensajes de corre=
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st data. Click to get it now.> http://sourceforge.net/powerbar/db2/> ______=
_________________________________________> opensipstack-devel mailing list>=
ope...@li...> https://lists.sourceforge.net/li=
sts/listinfo/opensipstack-devel
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e.
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|
|
From: Giuseppe B. <giu...@bs...> - 2007-07-11 14:19:44
|
Hi, I'm interested in IVR and VXML support, both present in the road map published on the web, but not present in your last list. Have you removed these features or are not related to OSSPhone? Thank you very much, Giuseppe Battaglia -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of = Ilian Jeri C. Pinzon Sent: marted=EC 10 luglio 2007 11.28 To: ope...@li... Subject: [OpenSIPStack] OSSPhone road map Hello Everyone, I'm planning to include the following features in OSSPhone. If you wish=20 some feature to be included or if you have any suggestions, please don't = hesitate to ask. Maybe I could make a room for it somewhere. :) Thanks! =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D OSSPhone road map 1. Call transfer and call hold (July 27) 2. USB phone support (August) 3. Mac support (3Q) 4. Multiple call lines (3Q) 5. IM support (3Q) 6. Playing of audio files over the wire (4Q) 7. Video support (4Q) Regards, Ilian -------------------------------------------------------------------------= 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 |
|
From: tomach <to...@dg...> - 2007-07-10 12:53:02
|
Hello! I just find small problme but if you are interested.. just tested that Trying event do not come... |
|
From: dial2world d. <dia...@ho...> - 2007-07-10 10:56:33
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Hi Ilian., Thanks for the update and nice to know that Alt sip will be in line with other leading softphone dialers in the market implementing state of the art features. We will be more than glad to be of assistance in whichever way we can and expedite especially the USB phone support (August). Currently we have shortlisted two vendors i.e a) http://www.tclcomm.cn/product_detail.asp?TypeID=12&Pid=10149 http://www.tclcomm.cn/product_detail.asp?TypeID=12&Pid=10128 (We have already the usb gateways with us for testing and we can also request the vendors engineers for interopability assistance with our softphone) b) http://yealink.com/en/view.asp?ClassLayer=17 (We have the SDK of the vendor and if required the vendors are ready for any assistance) Alternatively if you have shortlisted any other vendor kindly let us know.. How about "conferencing"??Does it include as you have stated Multiple call lines. Thanks & regards Salish >From: "Ilian Jeri C. Pinzon" <ip...@so...> >Reply-To: ope...@li... >To: ope...@li... >Subject: [OpenSIPStack] OSSPhone road map >Date: Tue, 10 Jul 2007 17:28:25 +0800 > >Hello Everyone, > >I'm planning to include the following features in OSSPhone. If you wish >some feature to be included or if you have any suggestions, please don't >hesitate to ask. Maybe I could make a room for it somewhere. :) > >Thanks! > >============================================== > >OSSPhone road map > >1. Call transfer and call hold (July 27) >2. USB phone support (August) >3. Mac support (3Q) >4. Multiple call lines (3Q) >5. IM support (3Q) >6. Playing of audio files over the wire (4Q) >7. Video support (4Q) > >Regards, >Ilian > > >------------------------------------------------------------------------- >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 _________________________________________________________________ Spice up your IM conversations. New, colorful and animated emoticons. Get chatting! http://server1.msn.co.in/SP05/emoticons/ |
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From: Andre S. <eds...@ya...> - 2007-07-10 09:52:53
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Could you add conference rooms. So we could put barging and call conference "Ilian Jeri C. Pinzon" <ip...@so...> wrote: Hello Everyone, I'm planning to include the following features in OSSPhone. If you wish some feature to be included or if you have any suggestions, please don't hesitate to ask. Maybe I could make a room for it somewhere. :) Thanks! ============================================== OSSPhone road map 1. Call transfer and call hold (July 27) 2. USB phone support (August) 3. Mac support (3Q) 4. Multiple call lines (3Q) 5. IM support (3Q) 6. Playing of audio files over the wire (4Q) 7. Video support (4Q) Regards, Ilian ------------------------------------------------------------------------- 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 --------------------------------- Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. |
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From: Ilian J. C. P. <ip...@so...> - 2007-07-10 09:28:37
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Hello Everyone, I'm planning to include the following features in OSSPhone. If you wish some feature to be included or if you have any suggestions, please don't hesitate to ask. Maybe I could make a room for it somewhere. :) Thanks! ============================================== OSSPhone road map 1. Call transfer and call hold (July 27) 2. USB phone support (August) 3. Mac support (3Q) 4. Multiple call lines (3Q) 5. IM support (3Q) 6. Playing of audio files over the wire (4Q) 7. Video support (4Q) Regards, Ilian |
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From: Ilian J. C. P. <ip...@so...> - 2007-07-09 12:24:55
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Hi, sebastian pastor wrote: > Hi all, > > 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) > You can check the sample programs in ATLSIP\OSSPhone\*. For the C++ .NET version (in OSSPhone\vc80\), you can look at Form1.h. For the C++ MFC version (in OSSPhone\vc80-mfc\, OSSPhone.cpp and OSSPhoneDlg.cpp. If you also intend to look at the phone's behavior at a much lower level, you can check out at the OpenSIPStack project. SoftPhone.cxx, SoftPhoneInterface.cxx would be a good place to start. Regards, Ilian > > > > > 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 > > > |
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From: Ilian J. C. P. <ip...@so...> - 2007-07-09 11:57:18
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Hi Salish, Support for USB Phones is in the roadmap for the SF. Will definitely implement this in the future. Please wait for updates in this list. Will also consider your offer regarding testing. Thank you very much. Regards, Ilian dial2world dl2 wrote: > > 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? >> > >> > >> ------------------------------------------------------------------------- >> >> > 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 > > _________________________________________________________________ > Voice your questions and our experts will answer them > http://content.msn.co.in/Lifestyle/AskExpert/Default01.htm > > > |
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