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From: Ilian J. C. P. <ip...@so...> - 2007-07-03 05:17:21
|
Hi, Sorry if it was not clear. I meant I already checked-in a fix for your problem. The changes are in OpalOSSConnection.cxx of OpenSIPStack. Could you check if this solved it? Thanks. Regards, Ilian tomach wrote: > I tested both scenarios I described and results are also in my previous message. What would Youlike me to test now? I do nto understand.... > > ------------------------------------------------------------------------- > 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...> - 2007-07-02 11:31:12
|
Hello , I found a bug on messaging using OpenSBC. It only recognizes 4 character account number. Meaning if the account is less than 4 characters it cannot receive chat message but it can send to a 4 character or higher account numbers. Example that cannot receive messages: sip:1...@ta... or sip:12...@ta... or sip:12...@ta... Could you check why? Thanks --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing. |
|
From: tomach <to...@dg...> - 2007-07-02 11:13:16
|
I tested both scenarios I described and results are also in my previous message. What would Youlike me to test now? I do nto understand.... |
|
From: Ilian J. C. P. <ip...@so...> - 2007-07-02 10:33:37
|
Btw, the mod is in OpenSIPStack. Specifically, OpalOSSConnection.cxx. Ilian Jeri C. Pinzon wrote: > Hi Tom, > > I've checked in a fix for this issue. Can you test if this works in > your scenario? > > Thanks. > > Regards, > Ilian > > tomach wrote: >> Hello! >> >> Yes it is . >> >> I localized another problem. Scenario: >> On sipserver i have this order of codecs: >> G.711 A >> G.729 >> >> On ATLSIP : >> G.729 >> G.711 A >> >> After SipServer send invite to ATLSIP it responses with OK and in SDP >> there are two codecs choosen!!! (should be only one). There is: >> G. 711 A >> G. 729 >> >> In ATLSIP I hear some noices all the time, i belive that codec then >> is uncorect (finally rtp is using g.711A I sniffed it with ethereal). >> >> Whne I call other way: from ATLSIP to SIPSErver it works correct. >> always the first in ATLSIP is choosen. >> >> Concluding I belive there are problmes when codecs are set between >> sipserver and atlsip. (And it seems that ATLSIP has problems wiht >> choosing codec). >> Scenario2: >> >> On sipserver i have this order of codecs: >> G.711 A >> >> On ATLSIP : >> G.729 >> G.711 A >> >> After SipServer send invite to ATLSIP it responses with OK and in SDP >> there is only one codec : >> G. 711 A >> >> But this time everything works correct. G711 A is chooosen and >> everythig works correct. >> >> >> Did you notice this problem before? >> >> ------------------------------------------------------------------------- >> >> 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...> - 2007-07-02 10:16:12
|
Hi Tom, I've checked in a fix for this issue. Can you test if this works in your scenario? Thanks. Regards, Ilian tomach wrote: > Hello! > > Yes it is . > > I localized another problem. Scenario: > On sipserver i have this order of codecs: > G.711 A > G.729 > > On ATLSIP : > G.729 > G.711 A > > After SipServer send invite to ATLSIP it responses with OK and in SDP there are two codecs choosen!!! (should be only one). There is: > G. 711 A > G. 729 > > In ATLSIP I hear some noices all the time, i belive that codec then is uncorect (finally rtp is using g.711A I sniffed it with ethereal). > > Whne I call other way: from ATLSIP to SIPSErver it works correct. always the first in ATLSIP is choosen. > > Concluding I belive there are problmes when codecs are set between sipserver and atlsip. (And it seems that ATLSIP has problems wiht choosing codec). > > Scenario2: > > On sipserver i have this order of codecs: > G.711 A > > On ATLSIP : > G.729 > G.711 A > > After SipServer send invite to ATLSIP it responses with OK and in SDP there is only one codec : > G. 711 A > > But this time everything works correct. G711 A is chooosen and everythig works correct. > > > Did you notice this problem before? > > ------------------------------------------------------------------------- > 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...> - 2007-07-02 07:53:36
|
Hello! Yes it is . I localized another problem. Scenario: On sipserver i have this order of codecs: G.711 A G.729 On ATLSIP : G.729 G.711 A After SipServer send invite to ATLSIP it responses with OK and in SDP there are two codecs choosen!!! (should be only one). There is: G. 711 A G. 729 In ATLSIP I hear some noices all the time, i belive that codec then is uncorect (finally rtp is using g.711A I sniffed it with ethereal). Whne I call other way: from ATLSIP to SIPSErver it works correct. always the first in ATLSIP is choosen. Concluding I belive there are problmes when codecs are set between sipserver and atlsip. (And it seems that ATLSIP has problems wiht choosing codec). Scenario2: On sipserver i have this order of codecs: G.711 A On ATLSIP : G.729 G.711 A After SipServer send invite to ATLSIP it responses with OK and in SDP there is only one codec : G. 711 A But this time everything works correct. G711 A is chooosen and everythig works correct. Did you notice this problem before? |
|
From: Joegen E. B. <joe...@gm...> - 2007-07-01 00:25:13
|
Make sure the format of your wave file is 16 bit 8 KHZ Mono.
tomach wrote:
> I did like this and I hear really weird sound in speakers :(
>
> PFilePath pf("C:\ringin.wav");
> m_RingBackToneFile = new MS::VoiceFile();
> m_RingBackToneFile->Open(pf, MS::VoiceFile::Wav_8khz_Mono,"RING_XXX" );
>
> Any ideas?
>
> I guess I could use totally outside livbrary to play wave when event is fired but i owudl prefer to keep it in one place and use one activex....
>
|
|
From: tomach <to...@dg...> - 2007-06-29 13:42:59
|
I did like this and I hear really weird sound in speakers :(
PFilePath pf("C:\ringin.wav");
m_RingBackToneFile = new MS::VoiceFile();
m_RingBackToneFile->Open(pf, MS::VoiceFile::Wav_8khz_Mono,"RING_XXX" );
Any ideas?
I guess I could use totally outside livbrary to play wave when event is fired but i owudl prefer to keep it in one place and use one activex....
|
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-29 09:58:09
|
The attachment was not sent. Probably filtered by the server. Anyway, here it is: =========================================================================== Hi all, I have exposed the setting of silence detection mode and audio jitter delay in ATLSIP and SoftPhoneInterface. Here are the methods: DisableSilenceDetection() - Disables silence detection. Disables CNG as well. EnableFixedSilenceDetection( ULONG threshold ) - Enables fixed silence detection. Any sound level below the threshold is treated as silence (and CN is generated as a result). Don't use too high threshold values or you'll only hear comfort noise. Try threshold=3 as suggested by Whit in another thread. EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, ULONG silenceDeadband ) - An extended version of the previous method. Don't tinker with this unless you know what you're doing. For reference on how signalDeadband and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket(). EnableAdaptiveSilenceDetection( ULONG adaptivePeriod ) - Enables an adaptive silence detection. Supposedly this enables the threshold to *adapt* to the current sound level every adaptivePeriod milliseconds. However, its silence detection doesn't seem to be very effective (at least in my machine). I'll look into this further to see what's wrong. This mode with adaptivePeriod=4800 is the default mode for ATLSIP. EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG signalDeadband, ULONG silenceDeadband ) - An extended version of the previous method. Don't tinker with this unless you know what you're doing. For reference on how signalDeadband and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket(). SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay ) - Sets audio jitter delay settings. Regards, Ilian Ilian Jeri C. Pinzon wrote: > ATLSIP's default silence detection is still faulty. You can try > setting the silence detection to fixed mode. Somebody said in another > thread that a threshold of 3 will do the trick. You may have to tweak > this value though. > > Attached is the thread containing the functions to call for setting > the silence detection mode. > > - Ilian > > tomach wrote: >> Hello! >> >> Ok thank you very much! Till now everythign works correct :) Now some >> more advanced test...about quality of voice etc... >> >> It started to work correctly (From Tag was ok) after I build it as >> debug and reregistered again :) Now everythign works correct:) >> >> I still have like last question, do you have comfort noise generator? >> Because now its totally silence if somoebdy do not talk at all... and >> it could be uncomcofortable for subscirbers... >> >> Best Regards, >> Tom >> >> ------------------------------------------------------------------------- >> >> 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 > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.476 / Virus Database: 269.9.12/878 - Release Date: 6/28/2007 5:57 PM > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-29 09:57:23
|
The attachment was not sent. Probably filtered by the server. Anyway, here it is: =========================================================================== Hi all, I have exposed the setting of silence detection mode and audio jitter delay in ATLSIP and SoftPhoneInterface. Here are the methods: DisableSilenceDetection() - Disables silence detection. Disables CNG as well. EnableFixedSilenceDetection( ULONG threshold ) - Enables fixed silence detection. Any sound level below the threshold is treated as silence (and CN is generated as a result). Don't use too high threshold values or you'll only hear comfort noise. Try threshold=3 as suggested by Whit in another thread. EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, ULONG silenceDeadband ) - An extended version of the previous method. Don't tinker with this unless you know what you're doing. For reference on how signalDeadband and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket(). EnableAdaptiveSilenceDetection( ULONG adaptivePeriod ) - Enables an adaptive silence detection. Supposedly this enables the threshold to *adapt* to the current sound level every adaptivePeriod milliseconds. However, its silence detection doesn't seem to be very effective (at least in my machine). I'll look into this further to see what's wrong. This mode with adaptivePeriod=4800 is the default mode for ATLSIP. EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG signalDeadband, ULONG silenceDeadband ) - An extended version of the previous method. Don't tinker with this unless you know what you're doing. For reference on how signalDeadband and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket(). SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay ) - Sets audio jitter delay settings. Regards, Ilian Ilian Jeri C. Pinzon wrote: > ATLSIP's default silence detection is still faulty. You can try > setting the silence detection to fixed mode. Somebody said in another > thread that a threshold of 3 will do the trick. You may have to tweak > this value though. > > Attached is the thread containing the functions to call for setting > the silence detection mode. > > - Ilian > > tomach wrote: >> Hello! >> >> Ok thank you very much! Till now everythign works correct :) Now some >> more advanced test...about quality of voice etc... >> >> It started to work correctly (From Tag was ok) after I build it as >> debug and reregistered again :) Now everythign works correct:) >> >> I still have like last question, do you have comfort noise generator? >> Because now its totally silence if somoebdy do not talk at all... and >> it could be uncomcofortable for subscirbers... >> >> Best Regards, >> Tom >> >> ------------------------------------------------------------------------- >> >> 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 > > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.476 / Virus Database: 269.9.12/878 - Release Date: 6/28/2007 5:57 PM > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-29 09:54:33
|
ATLSIP's default silence detection is still faulty. You can try setting the silence detection to fixed mode. Somebody said in another thread that a threshold of 3 will do the trick. You may have to tweak this value though. Attached is the thread containing the functions to call for setting the silence detection mode. - Ilian tomach wrote: > Hello! > > Ok thank you very much! Till now everythign works correct :) Now some more advanced test...about quality of voice etc... > > It started to work correctly (From Tag was ok) after I build it as debug and reregistered again :) Now everythign works correct:) > > I still have like last question, do you have comfort noise generator? Because now its totally silence if somoebdy do not talk at all... and it could be uncomcofortable for subscirbers... > > Best Regards, > Tom > > ------------------------------------------------------------------------- > 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...> - 2007-06-29 09:47:08
|
Try this. Probably the easiest way to do this is to get a .wav file for your custom ring sound. Go to SoftPhoneInterface::SoftPhoneInterface() and replace the following lines static PDTMFEncoder rt; rt.GenerateRingBackTone(); m_RingBackToneFile = new MS::VoiceFile( rt, MS::VoiceFile::Wav_8khz_Mono, "RING_US" ); with m_RingBackToneFile = new MS::VoiceFile(); m_RingBackToneFile.Open( PATH_TO_WAV_FILE, MS:VoiceFile:Wav_8khz_Mono, "RING_XXX" ); - Ilian tomach wrote: > Hello! > > Is it possible to change ring sound/? I maen now when connection comes sound isnt like ring...can I change it to ring sound? > > or is it possible to add dialing sound to ATLSIP? > > ------------------------------------------------------------------------- > 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...> - 2007-06-29 09:29:34
|
Hello! Is it possible to change ring sound/? I maen now when connection comes sound isnt like ring...can I change it to ring sound? or is it possible to add dialing sound to ATLSIP? |
|
From: tomach <to...@dg...> - 2007-06-28 12:25:42
|
Hello! Ok thank you very much! Till now everythign works correct :) Now some more advanced test...about quality of voice etc... It started to work correctly (From Tag was ok) after I build it as debug and reregistered again :) Now everythign works correct:) I still have like last question, do you have comfort noise generator? Because now its totally silence if somoebdy do not talk at all... and it could be uncomcofortable for subscirbers... Best Regards, Tom |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-28 06:27:24
|
Hi, m_SIPDomain is set by calling SoftPhoneManager::SetSIPDomain() Particularly by m_SoftPhone->SetSIPDomain() in SoftPhoneInterface.cxx. Hmmm. I think I know where your problem lies... Please check-out Form1.h of OSSPhone. I've made modifications in btnMakeCall_Click() and see if this solves your problem. Regards, Ilian tomach wrote: > OK I localized my problem. Its in > > m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetDomain( m_SIPDomain ); > m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetRealm( m_SIPDomain ); > > I exchanged those lines with: > > m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetDomain( "192.168.2.69" ); > m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetRealm(:192.168.2.69" ); > > Now From tag looks good and everythign works fine, > > hmmm I was trying to find whre m_SIPDomian is set but can not find it:( is it set anywehrer? > > ------------------------------------------------------------------------- > 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...> - 2007-06-27 12:16:18
|
Kolneath SOMETH wrote: > > Hi Joegen, > > Just a general question regarding OpenSBC. As it is written in C++, is > it possible to embed the application into the embedded system ie. ADSL > box, Router,.. ? PWLIB, the underlying library for OpenSIPStack and OpenSBC is known to compile in embedded systems using a GCC tool-chain. I haven't tried it personally but this should be possible. I have to be honest that OpenSIPStack is not the library of choice for embedding. I would be suggesting an ANSI C implementation such as osip. PJSIP is also another stack geared towards embedding. http://www.pjsip.org/ > Can we implement the OpenSBC to use the stack library from other > vendors (different from OpenSIPStack) ? Can I ask the reason why you would want to use another stack? I don't think it would be practical or even doable to let OpenSBC use another stack at this point. In fact, OpenSBC is a mere wrapper for functionalities within OpenSIPStack. The B2BUA which is at the heart of OpenSBC is within the OpenSIPStack realm. > > Thanks n Regards, > Kolneath SOMETH > |
|
From: tomach <to...@dg...> - 2007-06-27 12:11:43
|
OK I localized my problem. Its in m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetDomain( m_SIPDomain ); m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetRealm( m_SIPDomain ); I exchanged those lines with: m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetDomain( "192.168.2.69" ); m_SIPEndPoint->GetProfile().GetRegistrationProfile().SetRealm(:192.168.2.69" ); Now From tag looks good and everythign works fine, hmmm I was trying to find whre m_SIPDomian is set but can not find it:( is it set anywehrer? |
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From: Joegen E. B. <joe...@gm...> - 2007-06-27 11:39:23
|
Ilian, __debugbreak(); would be another option. This used to be _asm int 3. This would force your app to barf intentionally. Ilian Jeri C. Pinzon wrote: > Hi Tom, > > I also can't break into that part of the code. I think this has to do > with the ActiveX control being an external process. What you can do > though is to put MessageBox() for printing variables you need to check. > Manual but still effective. :) > > Guys, > > Any of you know how to debug an ActiveX control? > > Regards, > Ilian > > tomach wrote: > >> Hmm everytime I set breakpiont in SoftPhoneManager::InitializeSIP() it is not valid. >> there is this kind of announcement: >> >> The breakpoint will not currently be hit. No symbols have been loaded for this document. >> >> I am quite new to visual studio so can you advice why i can not set breakpoint 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 >> >> >> >> > > > ------------------------------------------------------------------------- > 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 2513 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 |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-27 11:26:06
|
Hi Tom, I also can't break into that part of the code. I think this has to do with the ActiveX control being an external process. What you can do though is to put MessageBox() for printing variables you need to check. Manual but still effective. :) Guys, Any of you know how to debug an ActiveX control? Regards, Ilian tomach wrote: > Hmm everytime I set breakpiont in SoftPhoneManager::InitializeSIP() it is not valid. > there is this kind of announcement: > > The breakpoint will not currently be hit. No symbols have been loaded for this document. > > I am quite new to visual studio so can you advice why i can not set breakpoint 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...> - 2007-06-27 10:51:58
|
Hmm everytime I set breakpiont in SoftPhoneManager::InitializeSIP() it is not valid. there is this kind of announcement: The breakpoint will not currently be hit. No symbols have been loaded for this document. I am quite new to visual studio so can you advice why i can not set breakpoint there? |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-27 09:03:02
|
Also, did you re-register the ATLSIP control after compiling? Just in case.
Regards,
Ilian
Ilian Jeri C. Pinzon wrote:
> Hi Tom,
>
> Can you check the values of m_SIPDomain and m_ProxyAddress in
> SoftPhoneManager::InitializeSIP() before making a call?
>
> You can also look in the following lines of CallSession::MakeCall(). Try
> checking the value of localHost.
>
> PString localHost;
>
> PString domain =
> !m_SessionProfile.GetRegistrationProfile().GetDomain().IsEmpty() ?
> m_SessionProfile.GetRegistrationProfile().GetDomain() :
> m_SessionProfile.GetProxyProfile().GetDomain();
>
> if( !domain.IsEmpty() )
> localHost = domain;
> else if( willUseProxy )
> {
> localHost =
> m_SessionProfile.GetRegistrationProfile().GetServerAddress();
> if( localHost.IsEmpty() )
> {
> ///get the domain from the proxy address
> SIPURI uri( m_SessionProfile.GetProxyProfile().GetServerAddress() );
> localHost = uri.GetHost();
> }
> }
>
> if( localHost.IsEmpty() )
> {
> localHost = via.GetAddress();
> }
>
> userURI << localHost;
>
> Regards,
> Ilian
>
> tomach wrote:
>
>> OK.
>> I tried it with three different sipServers (different vendors) everytime ATLSample behave the same.
>> FROM field was not correct :(
>>
>> It is really weird that it works correct with your sipServer.
>>
>> Best Regards,
>> Tom
>>
>> -------------------------------------------------------------------------
>> 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...
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From: Ilian J. C. P. <ip...@so...> - 2007-06-27 09:00:33
|
Hi Tom,
Can you check the values of m_SIPDomain and m_ProxyAddress in
SoftPhoneManager::InitializeSIP() before making a call?
You can also look in the following lines of CallSession::MakeCall(). Try
checking the value of localHost.
PString localHost;
PString domain =
!m_SessionProfile.GetRegistrationProfile().GetDomain().IsEmpty() ?
m_SessionProfile.GetRegistrationProfile().GetDomain() :
m_SessionProfile.GetProxyProfile().GetDomain();
if( !domain.IsEmpty() )
localHost = domain;
else if( willUseProxy )
{
localHost =
m_SessionProfile.GetRegistrationProfile().GetServerAddress();
if( localHost.IsEmpty() )
{
///get the domain from the proxy address
SIPURI uri( m_SessionProfile.GetProxyProfile().GetServerAddress() );
localHost = uri.GetHost();
}
}
if( localHost.IsEmpty() )
{
localHost = via.GetAddress();
}
userURI << localHost;
Regards,
Ilian
tomach wrote:
> OK.
> I tried it with three different sipServers (different vendors) everytime ATLSample behave the same.
> FROM field was not correct :(
>
> It is really weird that it works correct with your sipServer.
>
> Best Regards,
> Tom
>
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From: rnones <fo...@op...> - 2007-06-27 01:29:42
|
hi devendra, can you give us the url on where did you try to download OpenSBC? thanks raymund > HI, > > I am Not able to Download the Open SBC . > The link is not working > > Regards > Devendra > > ------------------------------------------------------ > ------------------- > 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/opensipst > ack-devel |
|
From: tomach <to...@dg...> - 2007-06-26 13:42:07
|
OK. I tried it with three different sipServers (different vendors) everytime ATLSample behave the same. FROM field was not correct :( It is really weird that it works correct with your sipServer. Best Regards, Tom |
|
From: tomach <to...@dg...> - 2007-06-26 13:11:51
|
I was using both sample app and ossphone. I didnt code anything jsut run sample and try to make call. then i exmaine that everytime for FROM it took hostIP and it shoudl take domain and not my host ip..... Im trying to find where to change it but i guess it will take time....Its really weird that you can not simulate this situation.... As i said even in SampleATL i just write sip:8726@192.168.2.69 it sends invite where FROM is sip:192.168.2.45 where is should be 2.69.... weird that it only happend in my sample... |
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