Thread: Re: [OpenSIPStack] Problem with OnOutgoingCallConnected (Page 3)

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Hey!

OK everytihgn works great!

Thank you!

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?

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?
>
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Thanks it worked :)

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

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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>   

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.
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> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
> >
> >
>
>
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_________________________________________________________________
Voice your questions and our experts will answer them 
http://content.msn.co.in/Lifestyle/AskExpert/Default01.htm

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
>
>
>

Hello!

I just find small problme but if you are interested.. just tested that Trying event do not come...

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

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
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>
>
>   

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?

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?
>
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>
>
>   

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...

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...
>
> -------------------------------------------------------------------------
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> Download DB2 Express C - the FREE version of DB2 express and take
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> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Thanks a lot. To be honest I am not very good in C++ and COM objects...I hoped more that You tell me some easy way how to send all strings logs as an event to ATLSIP, that in C# I could catch it(log) and send it to my own log system....Because all my log systems are in C#...

I mean I would like to have a log system similar to sipMessage event is it possible with not small pice of work and it will not influence on stableness?

Or how to send logs to syslog? Have You consider such a solution? How do You cope with big log files finallly if You do not send them to syslog? I belive that after several days of working logs files can be really huge?

Hi,

tomach wrote:
> Or how to send logs to syslog? Have You consider such a solution? 
For syslog you can do this:

    Logger::SetDefaultLevel( 5 );

    static unsigned int options =
      LogDateAndTime |
      LogThread |
      LogAppendToFile;

    Logger::SetDefaultLogStream( new LoggingSystemLogStream(
      "ATLSIP", PIPSocket::Address("127.0.0.1"), 
PIPSocket::Address("127.0.0.1"), 514, 0, options ) );
> How do You cope with big log files finallly if You do not send them to syslog? I belive that after several days of working logs files can be really huge?
>   
Typically *for* ATLSIP, we really not need to store all the logs. If a 
bug/problem is found, you would want the logs only for that scenario. So 
storing the logs per application runtime shouldn't hurt. Of course, 
there is no problem if you prefer to keep logs for a longer period. 
You'll just have to implement your own housekeeping.

Regards,
Ilian

> -------------------------------------------------------------------------
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>   

Hello Ilian!

I have found next problem. 
Event ringing in ATLsip comes to often. It sends its twice during connection:
1.when sipserver sends message "180 Rinnging", atlsip fires event ringing,
2. when sipserver sends message "183 sessoin progress", atlsip firesevent ringing agaiin...

I belive that it shoudl fire ringing event only once??

I believe this behavior is just appropriate.  There might be other 
applications that might want to known if multiple provisional responses 
are received.   If you don't need the event, then just ignore it on your 
application.

Also, the subject in this thread is no longer the subject of this mail.  
Please change your subject to be more descriptive of the post content.

tomach wrote:
> Hello Ilian!
>
> I have found next problem. 
> Event ringing in ATLsip comes to often. It sends its twice during connection:
> 1.when sipserver sends message "180 Rinnging", atlsip fires event ringing,
> 2. when sipserver sends message "183 sessoin progress", atlsip firesevent ringing agaiin...
>
> I belive that it shoudl fire ringing event only once??
>
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>
>   

Thanks Ilian!!

Everythign works correct with syslog, but what about opnesipstack? is it also possisble to direct it to syslog or only ATLSIP can be directed to syslog?

best regards!

Hi Tom,

The logs you redirected to syslog actually came from OpenSIPStack using 
the PTRACE and LOG macros. ATLSIP is merely a wrapper for OpenSIPStack 
after all.

Regards,
Ilian

tomach wrote:
> Thanks Ilian!!
>
> Everythign works correct with syslog, but what about opnesipstack? is it also possisble to direct it to syslog or only ATLSIP can be directed to syslog?
>
> best regards!
>
> -------------------------------------------------------------------------
> 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
>
>
>   

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
>
>
>   

Speech API is only needed by the IVR portion of the code and hsould never affect ATLSIP functionality in any way<br /><br />Joegen<br />