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From: Joegen E. B. <joe...@gm...> - 2007-08-16 02:58:54
|
Andre, Tons of information can be found in the logs. Have you even tried looking at it? What do you see? Andre Silo wrote: > Julian, > > Have you tried calling from a remote phone connecting to OSBC? I tested it but after two rings on the local account the line just got disconnected. Is there a parameter that extends the number of rings before it gets rejected? > > "Julian F. Tasis, III" <jf...@id...> wrote: > Hi, > > Thanks for the reply. I transfer the call back to the IVR when it is > rejected and just added some internal header and it worked. > > By the way I modify the IVRTransferCall a little bit so that when the > media server is routing the call to B's destination, B will see the > profile of the caller and not the media server. > > May I know how did you record all the wav files OpenSBC is using? I tried > recording my own wav files but I can't hear my recordings hehe. > > Regards, > Julian F. Tasis, III > > Re: [OpenSIPStack] IVR handling on transfer call rejection > From: - 2007-08-15 03:30 > There is more work to be done before you will be able to do this. Ryan > is working on improvements to the API to allow the transfered > connection to reconnect to the IVR once the outbound call is > disconnected. Watch the updates. > > > Julian F. Tasis, III wrote: > >> Good day, >> >> Hi, I was playing around with the IVRHandler lately and getting to know >> how it works. Just got a problem when transferring a call to B >> destination. If the B rejects the call, I can't notify A that the call is >> dropped. I wanted to play also a specific speil to A to notify it. I tried >> getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but >> with no success. Is there any other way around to do that? >> >> >> Regards, >> Julian F. Tasis, III >> > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > --------------------------------- > Pinpoint customers who are looking for what you sell. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
|
From: Joegen E. B. <joe...@gm...> - 2007-08-16 02:55:17
|
Julian, Format should be 8khz 16-bit mono. Andre Silo wrote: > Just use the Windows recorder found on Accessories > > "Julian F. Tasis, III" <jf...@id...> wrote: Hi, > > Thanks for the reply. I transfer the call back to the IVR when it is > rejected and just added some internal header and it worked. > > By the way I modify the IVRTransferCall a little bit so that when the > media server is routing the call to B's destination, B will see the > profile of the caller and not the media server. > > May I know how did you record all the wav files OpenSBC is using? I tried > recording my own wav files but I can't hear my recordings hehe. > > Regards, > Julian F. Tasis, III > > Re: [OpenSIPStack] IVR handling on transfer call rejection > From: - 2007-08-15 03:30 > There is more work to be done before you will be able to do this. Ryan > is working on improvements to the API to allow the transfered > connection to reconnect to the IVR once the outbound call is > disconnected. Watch the updates. > > > Julian F. Tasis, III wrote: > >> Good day, >> >> Hi, I was playing around with the IVRHandler lately and getting to know >> how it works. Just got a problem when transferring a call to B >> destination. If the B rejects the call, I can't notify A that the call is >> dropped. I wanted to play also a specific speil to A to notify it. I tried >> getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but >> with no success. Is there any other way around to do that? >> >> >> Regards, >> Julian F. Tasis, III >> > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > --------------------------------- > Be a better Heartthrob. Get better relationship answers from someone who knows. > Yahoo! Answers - Check it out. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
|
From: Andre S. <eds...@ya...> - 2007-08-16 01:44:30
|
Julian, Have you tried calling from a remote phone connecting to OSBC? I tested it but after two rings on the local account the line just got disconnected. Is there a parameter that extends the number of rings before it gets rejected? "Julian F. Tasis, III" <jf...@id...> wrote: Hi, Thanks for the reply. I transfer the call back to the IVR when it is rejected and just added some internal header and it worked. By the way I modify the IVRTransferCall a little bit so that when the media server is routing the call to B's destination, B will see the profile of the caller and not the media server. May I know how did you record all the wav files OpenSBC is using? I tried recording my own wav files but I can't hear my recordings hehe. Regards, Julian F. Tasis, III Re: [OpenSIPStack] IVR handling on transfer call rejection From: - 2007-08-15 03:30 There is more work to be done before you will be able to do this. Ryan is working on improvements to the API to allow the transfered connection to reconnect to the IVR once the outbound call is disconnected. Watch the updates. Julian F. Tasis, III wrote: > Good day, > > Hi, I was playing around with the IVRHandler lately and getting to know > how it works. Just got a problem when transferring a call to B > destination. If the B rejects the call, I can't notify A that the call is > dropped. I wanted to play also a specific speil to A to notify it. I tried > getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but > with no success. Is there any other way around to do that? > > > Regards, > Julian F. Tasis, III ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Pinpoint customers who are looking for what you sell. |
|
From: Andre S. <eds...@ya...> - 2007-08-16 01:25:04
|
Just use the Windows recorder found on Accessories "Julian F. Tasis, III" <jf...@id...> wrote: Hi, Thanks for the reply. I transfer the call back to the IVR when it is rejected and just added some internal header and it worked. By the way I modify the IVRTransferCall a little bit so that when the media server is routing the call to B's destination, B will see the profile of the caller and not the media server. May I know how did you record all the wav files OpenSBC is using? I tried recording my own wav files but I can't hear my recordings hehe. Regards, Julian F. Tasis, III Re: [OpenSIPStack] IVR handling on transfer call rejection From: - 2007-08-15 03:30 There is more work to be done before you will be able to do this. Ryan is working on improvements to the API to allow the transfered connection to reconnect to the IVR once the outbound call is disconnected. Watch the updates. Julian F. Tasis, III wrote: > Good day, > > Hi, I was playing around with the IVRHandler lately and getting to know > how it works. Just got a problem when transferring a call to B > destination. If the B rejects the call, I can't notify A that the call is > dropped. I wanted to play also a specific speil to A to notify it. I tried > getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but > with no success. Is there any other way around to do that? > > > Regards, > Julian F. Tasis, III ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. |
|
From: Julian F. T. I. <jf...@id...> - 2007-08-15 13:19:19
|
Hi, Thanks for the reply. I transfer the call back to the IVR when it is rejected and just added some internal header and it worked. By the way I modify the IVRTransferCall a little bit so that when the media server is routing the call to B's destination, B will see the profile of the caller and not the media server. May I know how did you record all the wav files OpenSBC is using? I tried recording my own wav files but I can't hear my recordings hehe. Regards, Julian F. Tasis, III Re: [OpenSIPStack] IVR handling on transfer call rejection From: <joegen@op...> - 2007-08-15 03:30 There is more work to be done before you will be able to do this. Ryan is working on improvements to the API to allow the transfered connection to reconnect to the IVR once the outbound call is disconnected. Watch the updates. Julian F. Tasis, III wrote: > Good day, > > Hi, I was playing around with the IVRHandler lately and getting to know > how it works. Just got a problem when transferring a call to B > destination. If the B rejects the call, I can't notify A that the call is > dropped. I wanted to play also a specific speil to A to notify it. I tried > getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but > with no success. Is there any other way around to do that? > > > Regards, > Julian F. Tasis, III |
|
From: Ilian J. C. P. <ip...@so...> - 2007-08-15 11:46:28
|
Hi bart, Sorry for the late response. I've added changes to the ossphone MFC project so you wouldn't have to deal with anything. It should build correctly with the default settings. Just get OpenSIPStack and ATLSIP code from CVS and put them in the same folder. Open the MFC project and then build. This should work fine. - Ilian bart wrote: > atlsip\OSSPhone\vc80-mfc. > > This project really looks good. I would like to base my project around this. Could someone send me a VC8 project with the following structure. > > Project Directory ( containing solution) > lib (containing opensipstack.lib + opensipstackd.lib) > include (containing opensipstack includes) > softphone (or project directory - containg VC generated files) > > This is default structure + lib and include directories. > > If this project contained the correct linker settings I will be on my way and will promise MaxBeers :-) > > Thanks > bart > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Ilian J. C. P. <ip...@so...> - 2007-08-15 10:30:13
|
Hey guys, I've made several changes on OSSPhone. You should now be able to build the OSSPhone project directly in its own solution file without first building ATLSIP separately. The solution file is on the OSSPhone folder. I've modified this to build and reference the interop dlls automatically before OSSPhone's build event. I've also added a much cleaner exit behavior. Also make sure you have the latest OpenSIPStack code. Get this from CVS. Update me if you have problems. - Ilian Yacine Auczone wrote: > I'm sure that there is something wrong since the compilation process works great and also the app works > I think that it should work on any x86 windows xp PC but i still have troubles on many computers. > did the developpers tested any ATLSIP.dll based softphone on other computer thene the workstation? > why the ossphone installer still not updated? > ossphone installer that uses an non updated version of ATLSIP.dll works great on any PC. > > > > >> Date: Tue, 14 Aug 2007 13:07:34 -0500 >> From: de...@wh... >> To: ope...@li... >> Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib >> >> >> >> I think I'm on the right track generating the Interop assemblies here but >> please correct me if I'm wrong: >> >> References: >> >> "COM Interop Part 1: C# Client Tutorial" >> >> http://msdn2.microsoft.com/en-us/library/aa645736(vs.71).aspx#vcwlkcomintero >> ppart1cclienttutorialanchor1 >> >> "Type Library Importer (Tlbimp.exe)" >> >> http://msdn2.microsoft.com/en-us/library/tt0cf3sx(VS.80).aspx >> >> "Generating Primary Interop Assemblies" >> >> http://msdn2.microsoft.com/en-us/library/dwe56e27(VS.80).aspx >> >> >> >> So I simply ran this command: >> >> C:\NEWSIP\atlsip\Release>tlbimp ATLSIP.dll /out:MyATLSIP.dll >> >> Subsequently running: >> >> C:\NEWSIP\atlsip\Release>Ildasm MyATLSIP.dll >> >> Will show the content (definitions) of this new Interop library. >> >> >> This process creates a new Interop assembly for ATLSIP.dll which can then be >> used in the project. So by my reasoning, this is how the original >> Interop.AxATLSIPLib files were/are created. >> >> I'm very new to .NET so this assembly stuff is a bit challenging to get my >> head around. >> >> If anyone sees an error in this approach let me know, otherwise perhaps it >> will help others! Look forward to feedback! >> >> Regards, >> >> Whit >> >> >> >> >> >> >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of Whit >> Thiele >> Sent: Tuesday, August 14, 2007 12:35 PM >> To: ope...@li... >> Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib >> >> >> >> Ilian, >> >> Where are the Interop dll's created? I've tried looking through the project >> settings but I can't see where they are being created. >> >> It seems (from the datestamps) that the Interop files are being copied from >> somewhere, not created. >> >> Whit >> >> >> <snip> >> Whit, >> >> I take back what I said earlier: "You should only add ATLSIP as a >> reference. VS will take care of the rest." For some reason, VS >> *sometimes* doesn't automatically load the AxInterop dll. You may have >> to manually add this. Make sure all the references to the *Interop dlls >> are up-to-date. >> >> Regards, >> Ilian >> >>> >>> >> <snip> >> >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > _________________________________________________________________ > Exprimez-vous : créez la page d'accueil qui vous ressemble avec Live.com. > http://www.live.com/getstarted > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: <jo...@op...> - 2007-08-15 03:30:00
|
There is more work to be done before you will be able to do this. Ryan is working on improvements to the API to allow the transfered connection to reconnect to the IVR once the outbound call is disconnected. Watch the updates. Julian F. Tasis, III wrote: > Good day, > > Hi, I was playing around with the IVRHandler lately and getting to know > how it works. Just got a problem when transferring a call to B > destination. If the B rejects the call, I can't notify A that the call is > dropped. I wanted to play also a specific speil to A to notify it. I tried > getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but > with no success. Is there any other way around to do that? > > > Regards, > Julian F. Tasis, III > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: <jo...@op...> - 2007-08-15 03:24:43
|
Responses are not routed. They just get matched to corresponding
CORE. In the scenario you have pasted below, the response was given to
the Proxy Core. This means that the request that created the
transaction was bound to a remote destination.
Andrew Pogrebennyk wrote:
> Hi,
>
> I am trying to understand how the responses are routed in OpenSBC.
> B2BUserAgent::ProcessEvent checks if there is a SIPMessage event,
> differentiates between different types of requests, then if event is not
> a request, event dispatcher checks if it already has a session for this
> response. Once session has been found, it calls
> B2BUAEndPoint::ProcessStackEvent. The latter does the following magic to
> handle the responses:
> if( !msg.IsRequest() )
> {
> OString meth = msg.GetCSeqMethod();
> if( meth == "SUBSCRIBE" ||
> meth == "NOTIFY" ||
> meth == "PUBLISH" )
> {
> ///this is a response
> SIPSession::GCRef ref = proxy->FindGCRefByCallId(
> msg.GetCallId() );
> if( ref != NULL )
> {
> proxy->ProcessStackEvent( ref, event );
> return;
> }
> }
> }
>
> So far so good. The switch loop inside the
> SIPSessionManager::ProcessStackEvent function calls OnIncomingSIPMessage:
>
> case SIPStackEvent::Message:
> LOG_CONTEXT( LogDetail(), eventObject->GetCallId(),
> "Event"
> << ": " << "---> Inbound - "
> <<
> ((SIPMessageArrival&)*eventObject).GetMessage().GetStartLine() );
>
> OnIncomingSIPMessage( (SIPMessageArrival&)*eventObject, session );
> break;
>
> OnIncomingSIPMessage to dispatch a valid response calls
> SIPSession::OnIncomingSIPMessage. It does the following thing:
> if( messageEvent.GetMessage().IsRequest() )
> {
> if( !messageEvent.GetMessage().IsAck() )
> {
> PWaitAndSignal lock( m_CurrentUASRequestMutex );
> m_CurrentUASRequest = messageEvent.GetMessage();
> }
> }
>
> I have kind of reached a deadlock here. What I want to know is how the
> response if actually delivered to the local transaction layer or remote
> proxy. We've seen a scenario where OpenBSC is serving as a frontend to
> PortaSIP and from it we know that responses are a subject to a check of
> Request-URI and To-URI to distinguish between local handling and relay.
> I just do not see where these checks are done. Thanks,
>
>
|
|
From: Joegen E. B. <joe...@gm...> - 2007-08-15 03:13:01
|
Ashish Khare wrote: > Hi Joegen, > Please reply for my below mail. > > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in detail > about how it is implementing the NAtting/ ALG functinality and how it will > handle the Media streams. No there isn't yet. Your best bet is to analyze the code and ask questions. > For Call Transfer, we will use the relay approach. Then you may not need to parse RTP > Also, if i want to just relay the Media packets, can you let me know the > algorithm you have applied in the openSBC. > RTP is parsed in RTPSession::OnReceiveData(). You may override this function to not parse anything and just return e_ProcessPacket for all cases. I haven't done this myself so i'm not sure what side effects there will be. If that happens, you are on your own. > Also, in openSBC product, is High Availability supported or it is in > roadmap > ? > It is planned but did not get a date yet in the roadmap. > > On 8/10/07, *Ashish Khare* <ash...@gm... > <mailto:ash...@gm...>> wrote: > > Hi Baclor, > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in > detail about how it is implementing the NAtting/ ALG functinality > and how it will handle the Media streams. > For Call Transfer, we will use the relay approach. > Also, if i want to just relay the Media packets, can you let me > know the algorithm you have applied in the openSBC. > > Also, in openSBC product, is High Availability supported or it is > in roadmap ? > > > On 8/10/07, *Joegen E. Baclor* <joe...@gm... > <mailto:joe...@gm...>> wrote: > > inline... > > > Ashish Khare wrote: > > Hi Baclor, > > This is still not clear to me. > > Lets take a example: > > Sip Client A is talking to Sip Client B through > Proxy/B2BUA(P) which > > handles only SIP signaling messages. > > Now in Call, Sip Client A is trasnferred to Sip CLient C and > now B and > > C are talking. > > But still they are abel to talk. > > Then how this case is different from yours. Can you please > elaborate > > and explain to me. > > There are two ways OpenSBC handles REFER. The default is to > relay the > REFER to the UA and let the UA do the transfer request. This > is ok > because the UA knows that there will be a change in the audio > session. > The second way (Local REFER) will not relay the REFER. > Instead OpenSBC > do the transfer. This leaves the other UA to not know that > the call is > actually transfered. If the transfer succeeded, a new media > would with > a different SSRC would have been created. If OpenSBC just > relays that, > the UA may reject the packets because the ssrc has already > changed. > > > > > > we are considering to build ALG. We have our own SIP stack ( > Proxy and > > B2BUA ) but we dont have RTP stack. Also we dont want to > parse the > > RTP stream. Just Rely it, based on source and destination IP and > > ports. Is this feasible ?. We are also exploring your openSBC > if we > > can used it. > > > > Of course this is feasible. You will have to change some > lines of code > in the media interface but it wont take much. Just post > questions > about the code if you need to clarify something. > > > > > > > > > > > > |
|
From: Ashish K. <ash...@gm...> - 2007-08-14 19:46:46
|
Hi Joegen, Please reply for my below mail. Thanks for the reply. Is there any design document about openSBC, which will tell me in detail about how it is implementing the NAtting/ ALG functinality and how it will handle the Media streams. For Call Transfer, we will use the relay approach. Also, if i want to just relay the Media packets, can you let me know the algorithm you have applied in the openSBC. Also, in openSBC product, is High Availability supported or it is in roadmap ? On 8/10/07, Ashish Khare <ash...@gm...> wrote: > > Hi Baclor, > > Thanks for the reply. > Is there any design document about openSBC, which will tell me in detail > about how it is implementing the NAtting/ ALG functinality and how it will > handle the Media streams. > For Call Transfer, we will use the relay approach. > Also, if i want to just relay the Media packets, can you let me know the > algorithm you have applied in the openSBC. > > Also, in openSBC product, is High Availability supported or it is in > roadmap ? > > > On 8/10/07, Joegen E. Baclor <joe...@gm...> wrote: > > > > inline... > > > > > > Ashish Khare wrote: > > > Hi Baclor, > > > This is still not clear to me. > > > Lets take a example: > > > Sip Client A is talking to Sip Client B through Proxy/B2BUA(P) which > > > handles only SIP signaling messages. > > > Now in Call, Sip Client A is trasnferred to Sip CLient C and now B and > > > C are talking. > > > But still they are abel to talk. > > > Then how this case is different from yours. Can you please elaborate > > > and explain to me. > > > > There are two ways OpenSBC handles REFER. The default is to relay the > > REFER to the UA and let the UA do the transfer request. This is ok > > because the UA knows that there will be a change in the audio session. > > The second way (Local REFER) will not relay the REFER. Instead OpenSBC > > do the transfer. This leaves the other UA to not know that the call is > > actually transfered. If the transfer succeeded, a new media would with > > > > a different SSRC would have been created. If OpenSBC just relays that, > > the UA may reject the packets because the ssrc has already changed. > > > > > > > > > > we are considering to build ALG. We have our own SIP stack ( Proxy and > > > > > B2BUA ) but we dont have RTP stack. Also we dont want to parse the > > > RTP stream. Just Rely it, based on source and destination IP and > > > ports. Is this feasible ?. We are also exploring your openSBC if we > > > can used it. > > > > > > > > Of course this is feasible. You will have to change some lines of code > > in the media interface but it wont take much. Just post questions > > about the code if you need to clarify something. > > > > > > > > > > > > > > > > > > > |
|
From: Yacine A. <yac...@ms...> - 2007-08-14 19:22:51
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I'm sure that there is something wrong since the compilation process works = great and also the app works I think that it should work on any x86 windows xp PC but i still have troub= les on many computers. did the developpers tested any ATLSIP.dll based softphone on other computer= thene the workstation? why the ossphone installer still not updated? ossphone installer that uses an non updated version of ATLSIP.dll works gre= at on any PC. > Date: Tue, 14 Aug 2007 13:07:34 -0500 > From: de...@wh... > To: ope...@li... > Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib >=20 >=20 >=20 > I think I'm on the right track generating the Interop assemblies here but > please correct me if I'm wrong: >=20 > References: >=20 > "COM Interop Part 1: C# Client Tutorial" >=20 > http://msdn2.microsoft.com/en-us/library/aa645736(vs.71).aspx#vcwlkcomint= ero > ppart1cclienttutorialanchor1 >=20 > "Type Library Importer (Tlbimp.exe)" >=20 > http://msdn2.microsoft.com/en-us/library/tt0cf3sx(VS.80).aspx >=20 > "Generating Primary Interop Assemblies" >=20 > http://msdn2.microsoft.com/en-us/library/dwe56e27(VS.80).aspx >=20 >=20 >=20 > So I simply ran this command: >=20 > C:\NEWSIP\atlsip\Release>tlbimp ATLSIP.dll /out:MyATLSIP.dll >=20 > Subsequently running: >=20 > C:\NEWSIP\atlsip\Release>Ildasm MyATLSIP.dll=20 >=20 > Will show the content (definitions) of this new Interop library. >=20 >=20 > This process creates a new Interop assembly for ATLSIP.dll which can then= be > used in the project. So by my reasoning, this is how the original > Interop.AxATLSIPLib files were/are created.=20 >=20 > I'm very new to .NET so this assembly stuff is a bit challenging to get m= y > head around.=20 >=20 > If anyone sees an error in this approach let me know, otherwise perhaps i= t > will help others! Look forward to feedback! >=20 > Regards, >=20 > Whit >=20 >=20 >=20 >=20 >=20 >=20 > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of Wh= it > Thiele > Sent: Tuesday, August 14, 2007 12:35 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib >=20 >=20 >=20 > Ilian, >=20 > Where are the Interop dll's created? I've tried looking through the proje= ct > settings but I can't see where they are being created. >=20 > It seems (from the datestamps) that the Interop files are being copied fr= om > somewhere, not created. >=20 > Whit >=20 >=20 > <snip> > Whit, >=20 > I take back what I said earlier: "You should only add ATLSIP as a=20 > reference. VS will take care of the rest." For some reason, VS=20 > *sometimes* doesn't automatically load the AxInterop dll. You may have=20 > to manually add this. Make sure all the references to the *Interop dlls=20 > are up-to-date. >=20 > Regards, > Ilian > > =20 > <snip> >=20 >=20 >=20 >=20 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >=20 >=20 >=20 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel _________________________________________________________________ Exprimez-vous : cr=E9ez la page d'accueil qui vous ressemble avec Live.com. http://www.live.com/getstarted= |
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From: Whit T. <de...@wh...> - 2007-08-14 18:07:52
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I think I'm on the right track generating the Interop assemblies here but please correct me if I'm wrong: References: "COM Interop Part 1: C# Client Tutorial" http://msdn2.microsoft.com/en-us/library/aa645736(vs.71).aspx#vcwlkcomintero ppart1cclienttutorialanchor1 "Type Library Importer (Tlbimp.exe)" http://msdn2.microsoft.com/en-us/library/tt0cf3sx(VS.80).aspx "Generating Primary Interop Assemblies" http://msdn2.microsoft.com/en-us/library/dwe56e27(VS.80).aspx So I simply ran this command: C:\NEWSIP\atlsip\Release>tlbimp ATLSIP.dll /out:MyATLSIP.dll Subsequently running: C:\NEWSIP\atlsip\Release>Ildasm MyATLSIP.dll Will show the content (definitions) of this new Interop library. This process creates a new Interop assembly for ATLSIP.dll which can then be used in the project. So by my reasoning, this is how the original Interop.AxATLSIPLib files were/are created. I'm very new to .NET so this assembly stuff is a bit challenging to get my head around. If anyone sees an error in this approach let me know, otherwise perhaps it will help others! Look forward to feedback! Regards, Whit -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Whit Thiele Sent: Tuesday, August 14, 2007 12:35 PM To: ope...@li... Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib Ilian, Where are the Interop dll's created? I've tried looking through the project settings but I can't see where they are being created. It seems (from the datestamps) that the Interop files are being copied from somewhere, not created. Whit <snip> Whit, I take back what I said earlier: "You should only add ATLSIP as a reference. VS will take care of the rest." For some reason, VS *sometimes* doesn't automatically load the AxInterop dll. You may have to manually add this. Make sure all the references to the *Interop dlls are up-to-date. Regards, Ilian > <snip> ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: Whit T. <de...@wh...> - 2007-08-14 17:36:02
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Ilian, Where are the Interop dll's created? I've tried looking through the project settings but I can't see where they are being created. It seems (from the datestamps) that the Interop files are being copied from somewhere, not created. Whit <snip> Whit, I take back what I said earlier: "You should only add ATLSIP as a reference. VS will take care of the rest." For some reason, VS *sometimes* doesn't automatically load the AxInterop dll. You may have to manually add this. Make sure all the references to the *Interop dlls are up-to-date. Regards, Ilian > <snip> |
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From: Whit T. <de...@wh...> - 2007-08-14 13:39:17
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Ilian, Thanks for the suggestions. I'm going to rebuild ATLSIP and test it out today. I'll post how it goes or what I discover. By the way, I've noticed that there is a TON of useful information in this forum. Is there any plan to consolidate this into an online wiki or something so that people could post tutorials, setups, experiences etc. Things which you wouldn't normally see or post in an email forum, but provides an invaluable resource to end users. Food for thought.. Whit -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Ilian Jeri C. Pinzon Sent: Tuesday, August 14, 2007 7:03 AM To: ope...@li... Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib Hi Yacine, Yacine Auczone wrote: > Hello Ilian, > I have sent the sip.log file, have you find something? > I'm still having crashes. > I can't tell directly from your logs what's causing your problem. And I still can't reproduce this, unfortunately. Can you do the following again? 1. Make sure that your OpenSIPStack code is up-to-date 2. Rebuild ATLSIP (clean all dlls first; atlsip.dll, *interop dlls) 3. Rebuild OSSPhone (you may have to manually add references to AxInterop and Interop dlls) Also, may I know a few more details how to bug occurs? Like how many calls before the crash, the length of the calls, the settings you set in OSSPhone, etc. Does this occur also in Debug build? If so, can your debugger attach to it and see where the crash comes from? Whit, I take back what I said earlier: "You should only add ATLSIP as a reference. VS will take care of the rest." For some reason, VS *sometimes* doesn't automatically load the AxInterop dll. You may have to manually add this. Make sure all the references to the *Interop dlls are up-to-date. Regards, Ilian > <snip> ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: Julian F. T. I. <jf...@id...> - 2007-08-14 13:10:06
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Hi Ryan, I'm trying to integrate OpenSBC with a TAPI compliant modem and use its media server to work as PBX. I'm having problem when I reject a call transfer. I can't notify the caller that B already rejected the call. I tried getting the b2buacall from leg1 and leg2 then play a file but with no success. Am I doing the right thing? I wanted to notify the caller by playing a speil. Thanks, Julian F. Tasis, III |
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From: Ilian J. C. P. <ip...@so...> - 2007-08-14 12:03:36
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Hi Yacine, Yacine Auczone wrote: > Hello Ilian, > I have sent the sip.log file, have you find something? > I'm still having crashes. > I can't tell directly from your logs what's causing your problem. And I still can't reproduce this, unfortunately. Can you do the following again? 1. Make sure that your OpenSIPStack code is up-to-date 2. Rebuild ATLSIP (clean all dlls first; atlsip.dll, *interop dlls) 3. Rebuild OSSPhone (you may have to manually add references to AxInterop and Interop dlls) Also, may I know a few more details how to bug occurs? Like how many calls before the crash, the length of the calls, the settings you set in OSSPhone, etc. Does this occur also in Debug build? If so, can your debugger attach to it and see where the crash comes from? Whit, I take back what I said earlier: "You should only add ATLSIP as a reference. VS will take care of the rest." For some reason, VS *sometimes* doesn't automatically load the AxInterop dll. You may have to manually add this. Make sure all the references to the *Interop dlls are up-to-date. Regards, Ilian > <snip> |
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From: Yacine A. <yac...@ms...> - 2007-08-14 09:44:52
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Hello Ilian, I have sent the sip.log file, have you find something? I'm still having crashes. > Date: Tue, 14 Aug 2007 13:21:10 +0800 > From: ip...@so... > To: ope...@li... > Subject: Re: [OpenSIPStack] ATLSIPLib and Interop.AxATLSIPLib >=20 > Hello Whit, >=20 > No such thing as a dumb question. :) >=20 > Whit Thiele wrote: > > > > This is a question related to the problems I've been having with the > > ntdll.dll crash. I just want to ensure that I am using the correct libr= aries > > and installing everything correctly. > > > > > > When I compile ATLSIP in Release, it generates the ATLSIP.dll > > > > When I compile ATLSIP in Release-Minimal mode, it generates the > > AxInterop.ATLSIPLib.1.0.dll file in the Release folder, hover its times= tamp > > is when I first downloaded ATLSIP out of CVS, so I'm assuming this isn'= t > > compiled, but copied. > > =20 > It is compiled but not always. The Interop DLLs are there for=20 > interoperability between .NET and Win32/ATL. So if there are no major=20 > structural changes to ATLSIP, they don't need to compiled. Just check if= =20 > your AxInterop.ATLSIPLib.1.0.dll and Interop.ATLSIPLib.1.0.dll of your=20 > Release and Interop folders have the same timestamps. >=20 > Also, one quirk I found with VS is that it doesn't always automatically=20 > copy Interop.ATLSIPLib.1.0.dll to the target folder. You may have to=20 > manually copy this to your Release folder. > > > > Which library should be used in a .NET app as a reference? Should you i= mport > > the ATLSIP.dll directly, or should you use the Interop library? > > =20 > You should only add ATLSIP as a reference. VS will take care of the rest. >=20 > - Ilian > > > > Hopefully there is no such thing as a 'dumb' question! > > > > > > Regards, > > > > Whit > > > > > > > > > > -----------------------------------------------------------------------= -- > > This SF.net email is sponsored by: Splunk Inc. > > Still grepping through log files to find problems? Stop. > > Now Search log events and configuration files using AJAX and a browser. > > Download your FREE copy of Splunk now >> http://get.splunk.com/ > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > =20 >=20 >=20 > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel _________________________________________________________________ Essayez Live.com, votre nouvelle page d'accueil ! Personnalisez-la en quelq= ues clics pour retrouver tout ce qui vous int=E9resse au m=EAme endroit. http://www.live.com/getstarted= |
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From: Andre S. <eds...@ya...> - 2007-08-14 07:48:18
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Ryan, I could do this in the routes table: sip[1800123456*] sip:5000@192.168.0.1 So that when an inbound call comes in it routes to the Media Server. Ryan Colobong <rco...@so...> wrote: Hello Andre, You can access the IVR of OpenSBC by calling the number specified in the Media Server in HTTP Admin configuration (default number is 5000). The default behavior of the IVR in OpenSBC is that it will ask you to enter a pin/password where it would be validated using the Local Domain Accounts (HTTP Admin). After this it would ask for a destination number which would be based on your OpenSBC Routes (HTTP Admin). regards, Andre Silo wrote: > Julian, > > Sorry to disturb you but are there any settings you made on OSBC admin to activate this IVR. Or is did you create a VXML file? > > "Julian F. Tasis, III" wrote: > Hi Andre, > > Sorry if I can't reply under your message my mail server does not allow me > to send message aside from my office gateway. > > On IVRHandle you will be asked to put your security code, that is the > password of the caller and is registered on local domain accounts. > > Regards, > Julian F. Tasis, III > Product Planning and Development > > Ideawurx Inc. > 7th Floor, V.A. Rufino Bldg., > 6784 Ayala Ave. > Makati City, Philippines 1226 > > Tel: +(632) 8111530 > Fax: +(632) 8189767 > > jf...@id... > http://www.ideawurx.com.ph > > Ideawurx is a telecommunications solutions provider that develops and > markets a range of customized computer telephony integration ( CTI ) > solutions. > > "Our IDEAS will WORK for YOU!" > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > --------------------------------- > Luggage? GPS? Comic books? > Check out fitting gifts for grads at Yahoo! Search. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. |
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From: Ryan C. <rco...@so...> - 2007-08-14 06:55:57
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Hello Andre, You can access the IVR of OpenSBC by calling the number specified in the Media Server in HTTP Admin configuration (default number is 5000). The default behavior of the IVR in OpenSBC is that it will ask you to enter a pin/password where it would be validated using the Local Domain Accounts (HTTP Admin). After this it would ask for a destination number which would be based on your OpenSBC Routes (HTTP Admin). regards, Andre Silo wrote: > Julian, > > Sorry to disturb you but are there any settings you made on OSBC admin to activate this IVR. Or is did you create a VXML file? > > "Julian F. Tasis, III" <jf...@id...> wrote: > Hi Andre, > > Sorry if I can't reply under your message my mail server does not allow me > to send message aside from my office gateway. > > On IVRHandle you will be asked to put your security code, that is the > password of the caller and is registered on local domain accounts. > > Regards, > Julian F. Tasis, III > Product Planning and Development > > Ideawurx Inc. > 7th Floor, V.A. Rufino Bldg., > 6784 Ayala Ave. > Makati City, Philippines 1226 > > Tel: +(632) 8111530 > Fax: +(632) 8189767 > > jf...@id... > http://www.ideawurx.com.ph > > Ideawurx is a telecommunications solutions provider that develops and > markets a range of customized computer telephony integration ( CTI ) > solutions. > > "Our IDEAS will WORK for YOU!" > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > --------------------------------- > Luggage? GPS? Comic books? > Check out fitting gifts for grads at Yahoo! Search. > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Andre S. <eds...@ya...> - 2007-08-14 06:28:21
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Julian, Sorry to disturb you but are there any settings you made on OSBC admin to activate this IVR. Or is did you create a VXML file? "Julian F. Tasis, III" <jf...@id...> wrote: Hi Andre, Sorry if I can't reply under your message my mail server does not allow me to send message aside from my office gateway. On IVRHandle you will be asked to put your security code, that is the password of the caller and is registered on local domain accounts. Regards, Julian F. Tasis, III Product Planning and Development Ideawurx Inc. 7th Floor, V.A. Rufino Bldg., 6784 Ayala Ave. Makati City, Philippines 1226 Tel: +(632) 8111530 Fax: +(632) 8189767 jf...@id... http://www.ideawurx.com.ph Ideawurx is a telecommunications solutions provider that develops and markets a range of customized computer telephony integration ( CTI ) solutions. "Our IDEAS will WORK for YOU!" ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search. |
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From: Julian F. T. I. <jf...@id...> - 2007-08-14 05:41:04
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Hi Andre, Sorry if I can't reply under your message my mail server does not allow me to send message aside from my office gateway. On IVRHandle you will be asked to put your security code, that is the password of the caller and is registered on local domain accounts. Regards, Julian F. Tasis, III Product Planning and Development Ideawurx Inc. 7th Floor, V.A. Rufino Bldg., 6784 Ayala Ave. Makati City, Philippines 1226 Tel: +(632) 8111530 Fax: +(632) 8189767 jf...@id... http://www.ideawurx.com.ph Ideawurx is a telecommunications solutions provider that develops and markets a range of customized computer telephony integration ( CTI ) solutions. "Our IDEAS will WORK for YOU!" |
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From: Ilian J. C. P. <ip...@so...> - 2007-08-14 05:21:08
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Hello Whit, No such thing as a dumb question. :) Whit Thiele wrote: > > This is a question related to the problems I've been having with the > ntdll.dll crash. I just want to ensure that I am using the correct libraries > and installing everything correctly. > > > When I compile ATLSIP in Release, it generates the ATLSIP.dll > > When I compile ATLSIP in Release-Minimal mode, it generates the > AxInterop.ATLSIPLib.1.0.dll file in the Release folder, hover its timestamp > is when I first downloaded ATLSIP out of CVS, so I'm assuming this isn't > compiled, but copied. > It is compiled but not always. The Interop DLLs are there for interoperability between .NET and Win32/ATL. So if there are no major structural changes to ATLSIP, they don't need to compiled. Just check if your AxInterop.ATLSIPLib.1.0.dll and Interop.ATLSIPLib.1.0.dll of your Release and Interop folders have the same timestamps. Also, one quirk I found with VS is that it doesn't always automatically copy Interop.ATLSIPLib.1.0.dll to the target folder. You may have to manually copy this to your Release folder. > > Which library should be used in a .NET app as a reference? Should you import > the ATLSIP.dll directly, or should you use the Interop library? > You should only add ATLSIP as a reference. VS will take care of the rest. - Ilian > > Hopefully there is no such thing as a 'dumb' question! > > > Regards, > > Whit > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Andre S. <eds...@ya...> - 2007-08-14 05:08:20
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Julian, How did you route the call to the IVRHandle first greeting? "Julian F. Tasis, III" <jf...@id...> wrote: Good day, Hi, I was playing around with the IVRHandler lately and getting to know how it works. Just got a problem when transferring a call to B destination. If the B rejects the call, I can't notify A that the call is dropped. I wanted to play also a specific speil to A to notify it. I tried getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but with no success. Is there any other way around to do that? Regards, Julian F. Tasis, III ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel --------------------------------- Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. |
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From: Julian F. T. I. <jf...@id...> - 2007-08-14 03:47:10
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Good day, Hi, I was playing around with the IVRHandler lately and getting to know how it works. Just got a problem when transferring a call to B destination. If the B rejects the call, I can't notify A that the call is dropped. I wanted to play also a specific speil to A to notify it. I tried getting leg1 and leg2 b2buaconnection and tried to used ivrplayfile but with no success. Is there any other way around to do that? Regards, Julian F. Tasis, III |
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