Thread: [OpenSIPStack] Problem with OnOutgoingCallConnected
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From: tomach <to...@dg...> - 2007-06-05 09:05:15
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Hello!<br /><br />I am using atlsip activex.<br />I noticed that when I try to create connection, even call isnt established yet I recieve OnOutgoingCallConnected event. Also when connection is esablished I recieved it second time. Did anyone have the same kind of problem? <br /><br />Next thing is when i run softphone based on atlsip activex it does not send any events. But when I run it without visual studio it works fine. Was that bug reported before?<br /><br />Last thing I want to ask is: how can I know when the line is busy? I tried OnLogSIPMessage but then I have to parse it etc... I thought more about somethign like OnLineBusy event, does this kind of solution exist allready?<br /><br />Best Regards!<br /> |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-05 09:17:39
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Hi, tomach wrote: > Hello!<br /><br />I am using atlsip activex.<br />I noticed that when I try to create connection, even call isnt established yet I recieve OnOutgoingCallConnected event. Also when connection is esablished I recieved it second time. Did anyone have the same kind of problem? <br /><br />Next thing is when i run softphone based on atlsip activex it does not send any events. But when I run it without visual studio it works fine. Was that bug reported before?<br /><br /> I'll try to reproduce your problem. I'll get back to you later if I find anything. > Last thing I want to ask is: how can I know when the line is busy? I tried OnLogSIPMessage but then I have to parse it etc... I thought more about somethign like OnLineBusy event, does this kind of solution exist allready? ATLSIP should throw Event_OutgoingCallRejected. Regards, Ilian > <br /><br />Best Regards!<br /> > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: tomach <to...@dg...> - 2007-06-05 11:11:00
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Hello!<br /><br />So I was listening all events and this is what I recived when line was busy:<br /><br />ringing event<br /><br />callconnected event<br /><br />calldisconnected event<br /><br />thats all what I think is missing is :<br /><br />trying event, rejected event, <br /><br />I also belive there shouldnt be callconnected event because there was call connected cos line was busy???<br /><br />Am I right, or maybe I take it worng, events?<br /><br /><br /> |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-05 11:19:55
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Hi. Can you post the sip logs? - Ilian tomach wrote: > Hello!<br /><br />So I was listening all events and this is what I recived when line was busy:<br /><br />ringing event<br /><br />callconnected event<br /><br />calldisconnected event<br /><br />thats all what I think is missing is :<br /><br />trying event, rejected event, <br /><br />I also belive there shouldnt be callconnected event because there was call connected cos line was busy???<br /><br />Am I right, or maybe I take it worng, events?<br /><br /><br /> > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-05 11:22:19
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Hi,
tomach wrote:
> Hello!<br /><br />I am using atlsip activex.<br />I noticed that when I try to create connection, even call isnt established yet I recieve OnOutgoingCallConnected event. Also when connection is esablished I recieved it second time. Did anyone have the same kind of problem? <br /><br />Next thing is when i run softphone based on atlsip activex it does not send any events. But when I run it without visual studio it works fine. Was that bug reported before?<br /><br />
Have you done this in Visual Studio?
1. After compiling ATLSIP, go to Tools -> ActiveX Control Test Container
2. In ActiveX Control Test Container, go to File -> Register Controls
3. Find ATLSIP.OpenSIPStackCtl and reregister it.
4. If this does not work or you can't find ATLSIP.OpenSIPStackCtl, try
unregistering and then registering ATLSIP.dll from your output folder.
Regards,
Ilian
> Last thing I want to ask is: how can I know when the line is busy? I tried OnLogSIPMessage but then I have to parse it etc... I thought more about somethign like OnLineBusy event, does this kind of solution exist allready?<br /><br />Best Regards!<br />
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by DB2 Express
> Download DB2 Express C - the FREE version of DB2 express and take
> control of your XML. No limits. Just data. Click to get it now.
> http://sourceforge.net/powerbar/db2/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
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From: tomach <to...@dg...> - 2007-06-05 13:48:52
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Hmmm all the time I used two dlls that were wrappers for ATLSIPLib.1.0. AxI= nterop and Interop that I downloaded if from your cvs. When I tried to comp= ile it, it gave me an error C2061: syntax error __RPC__in.... LIke it could= nt find definition of this __RPC__. Have no idea how to fix it ?:( <br />He= re I attached the logs which gave my softphone when line was busy<br /><br = />----------------25:45.709----------------<br />*** LISTENER STARTED *** 1= 27.0.0.1:5060<br /><br />----------------25:45.720----------------<br />***= LISTENER STARTED *** 192.168.2.48:5060 [*** DEFAULT LISTENER ***]<br /><br= />----------------25:45.727----------------<br />*** LISTENER STARTED *** = 192.168.44.46:5060<br /><br />----------------25:45.731----------------<br = />*** LISTENER STARTED *** 192.168.100.1:5060<br /><br />----------------25= :45.740----------------<br />*** LISTENER STARTED *** 192.168.174.1:5060<br= /><br />----------------25:46.008----------------<br />SEND: XOR=3D0 630 B= ytes to 192.168.2.111:5060:UDP (INVITE sip:8726@192.168.2.111 SIP/2.0) Inte= rface Address=3D192.168.2.48<br />INVITE sip:8726@192.168.2.111 SIP/2.0<br = />From: <sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcb= fe<br />To: sip:8726@192.168.2.111<br />Via: SIP/2.0/UDP 192.168.2.48:5060;= iid=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.168= .2.111;rport<br />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9ecc-c= d703c2fcbfe<br />Contact: <sip:192.168.2.48:5060><br />Max-Forwards: = 70<br />Content-Type: application/sdp<br />Content-Length: 205<br /><br />v= =3D0<br />o=3D- 1181050975 1181050975 IN IP4 192.168.2.48<br />s=3DOSS RTP = Session<br />c=3DIN IP4 192.168.2.48<br />t=3D0 0<br />m=3Daudio 5000 RTP/A= VP 101 18<br />a=3Drtpmap:101 telephone-event/8000<br />a=3Dfmtp:101 0-15<b= r />a=3Drtpmap:18 G729/8000<br /><br /><br /><br />----------------25:46.05= 7----------------<br />RCV: XOR=3D0 513 Bytes from RCVADDR: 192.168.2.111:R= CVPORT: 5060:UDP (SIP/2.0 100 Trying)<br />SIP/2.0 100 Trying<br />From:&nb= sp; <sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<br />To= : sip:8726@192.168.2.111;tag=3D1c260462125<br />Via: SIP/2.0/UDP 192.168.2.= 48:5060;iid=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr= =3D192.168.2.111<br />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9e= cc-cd703c2fcbfe<br />Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.60A.03= 4<br />Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, R= EFER, INFO, SUBSCRIBE, UPDATE<br />Supported: em, timer, replaces, path<br = />Content-Length: 0<br /><br /><br />----------------25:46.179-------------= ---<br />RCV: XOR=3D0 856 Bytes from RCVADDR: 192.168.2.111:RCVPORT: 5060:U= DP (SIP/2.0 183 Session Progress)<br />SIP/2.0 183 Session Progress<br />Fr= om: <sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<b= r />To: sip:8726@192.168.2.111;tag=3D1c260462125<br />Via: SIP/2.0/UDP 192.= 168.2.48:5060;iid=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-= addr=3D192.168.2.111<br />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-181= 0-9ecc-cd703c2fcbfe<br />Contact: <sip:23@192.168.2.111;user=3Dphone>= <br />Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.60A.034<br />Allow: R= EGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUB= SCRIBE, UPDATE<br />Supported: em, timer, replaces, path<br />Content-Type:= application/sdp<br />Content-Length: 256<br /><br />v=3D0<br />o=3DAudioco= desGW 260533184 260532796 IN IP4 192.168.2.111<br />s=3DPhone-Call<br />c= =3DIN IP4 192.168.2.111<br />t=3D0 0<br />m=3Daudio 6290 RTP/AVP 18 101<br = />a=3Drtpmap:18 g729/8000<br />a=3Dfmtp:18 annexb=3Dno<br />a=3Drtpmap:101 = telephone-event/8000<br />a=3Dfmtp:101 0-15<br />a=3Dptime:20<br />a=3Dsend= recv<br /><br /><br /><br />----------------26:16.529----------------<br />= RCV: XOR=3D0 560 Bytes from RCVADDR: 192.168.2.111:RCVPORT: 5060:UDP (SIP/2= .0 486 Busy Here)<br />SIP/2.0 486 Busy Here<br />From: <sip:192.1= 68.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<br />To: sip:8726@192.16= 8.2.111;tag=3D1c260462125<br />Via: SIP/2.0/UDP 192.168.2.48:5060;iid=3D1;b= ranch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.168.2.111<br= />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9ecc-cd703c2fcbfe<br = />Contact: <sip:23@192.168.2.111;user=3Dphone><br />Server: Audiocode= s-Sip-Gateway-Mediant 2000/v.4.60A.034<br />Allow: REGISTER, OPTIONS, INVIT= E, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE<br />Sup= ported: em, timer, replaces, path<br />Content-Length: 0<br /><br /><br />-= ---------------26:16.534----------------<br />SEND: XOR=3D0 398 Bytes to 19= 2.168.2.111:5060:UDP (ACK sip:8726@192.168.2.111 SIP/2.0) Interface Address= =3D192.168.2.48<br />ACK sip:8726@192.168.2.111 SIP/2.0<br />From: &l= t;sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<br />To: sip:= 8726@192.168.2.111;tag=3D1c260462125<br />Via: SIP/2.0/UDP 192.168.2.48:506= 0;iid=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.1= 68.2.111;rport<br />CSeq: 4711 ACK<br />Call-ID: 9ecf126d-d<br /> |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-06 11:43:34
|
Hi,
tomach wrote:
> Hmmm all the time I used two dlls that were wrappers for ATLSIPLib.1.0.=
AxInterop and Interop that I downloaded if from your cvs. When I tried t=
o compile it, it gave me an error C2061: syntax error __RPC__in.... LIke =
it couldnt find definition of this __RPC__.=20
Most likely this error comes from compiling the ATLSIPSample project=20
before compiling and registering ATLSIP.
Do this:
1. Compile the ATLSIP project
2. Go to Tools -> ActiveX Control Test Container
3. In ActiveX Control Test Container, go to File -> Register Controls
4. Find ATLSIP.OpenSIPStackCtl and reregister it.
5. If this does not work or you can't find ATLSIP.OpenSIPStackCtl, try
unregistering and then registering ATLSIP.dll from your output folder=
=2E
then try to compile ATLSIPSample again.
> Have no idea how to fix it ?:( <br />Here I attached the logs which gav=
e my softphone when line was busy<br /><br=20
Only Event_OutgoingCallRejected is fired with the current version of=20
ATLSIP. You need to get the latest version from CVS then recompile.
Regards,
Ilian
> />----------------25:45.709----------------<br />*** LISTENER STARTED *=
** 127.0.0.1:5060<br /><br />----------------25:45.720----------------<br=
/>*** LISTENER STARTED *** 192.168.2.48:5060 [*** DEFAULT LISTENER ***]<=
br /><br />----------------25:45.727----------------<br />*** LISTENER ST=
ARTED *** 192.168.44.46:5060<br /><br />----------------25:45.731--------=
--------<br />*** LISTENER STARTED *** 192.168.100.1:5060<br /><br />----=
------------25:45.740----------------<br />*** LISTENER STARTED *** 192.1=
68.174.1:5060<br /><br />----------------25:46.008----------------<br />S=
END: XOR=3D0 630 Bytes to 192.168.2
> .111:5060:UDP (INVITE sip:8726@192.168.2.111 SIP/2.0) Interface Addres=
s=3D192.168.2.48<br />INVITE sip:8726@192.168.2.111 SIP/2.0<br />From:&nb=
sp; <sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<br />=
To: sip:8726@192.168.2.111<br />Via: SIP/2.0/UDP 192.168.2.48:5060;iid=3D=
1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.168.2.1=
11;rport<br />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9ecc-cd7=
03c2fcbfe<br />Contact: <sip:192.168.2.48:5060><br />Max-Forwards: =
70<br />Content-Type: application/sdp<br />Content-Length: 205<br /><br /=
>v=3D0<br />o=3D- 1181050975 1181050975 IN IP4 192.168.2.48<br />s=3DOSS =
RTP Session<br />c=3DIN IP4 192.168.2.48<br />t=3D0 0<br />m=3Daudio 5000=
RTP/AVP 101 18<br />a=3Drtpmap:101 telephone-event/8000<br />a=3Dfmtp:10=
1 0-15<br />a=3Drtpmap:18 G729/8000<br /><br /><br /><br />--------------=
--25:46.057----------------<br />RCV: XOR=3D0 513 Bytes from RCVADDR: 192=
=2E168.2.111:RCVPORT: 5060:UDP (SIP/2.0 100 Trying)<br />SIP/2.0 100 Tryi=
ng
> <br />From: <sip:192.168.2.48>;tag=3D9ecf126ddaf81810907bc=
d703c2fcbfe<br />To: sip:8726@192.168.2.111;tag=3D1c260462125<br />Via: S=
IP/2.0/UDP 192.168.2.48:5060;iid=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907a=
cd703c2fcbfe;uas-addr=3D192.168.2.111<br />CSeq: 4711 INVITE<br />Call-ID=
: 9ecf126d-daf8-1810-9ecc-cd703c2fcbfe<br />Server: Audiocodes-Sip-Gatewa=
y-Mediant 2000/v.4.60A.034<br />Allow: REGISTER, OPTIONS, INVITE, ACK, CA=
NCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE<br />Supported: =
em, timer, replaces, path<br />Content-Length: 0<br /><br /><br />-------=
---------25:46.179----------------<br />RCV: XOR=3D0 856 Bytes from RCVAD=
DR: 192.168.2.111:RCVPORT: 5060:UDP (SIP/2.0 183 Session Progress)<br />S=
IP/2.0 183 Session Progress<br />From: <sip:192.168.2.48>;tag=
=3D9ecf126ddaf81810907bcd703c2fcbfe<br />To: sip:8726@192.168.2.111;tag=3D=
1c260462125<br />Via: SIP/2.0/UDP 192.168.2.48:5060;iid=3D1;branch=3Dz9hG=
4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.168.2.111<br=20
> />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9ecc-cd703c2fcbfe=
<br />Contact: <sip:23@192.168.2.111;user=3Dphone><br />Server: Aud=
iocodes-Sip-Gateway-Mediant 2000/v.4.60A.034<br />Allow: REGISTER, OPTION=
S, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDAT=
E<br />Supported: em, timer, replaces, path<br />Content-Type: applicatio=
n/sdp<br />Content-Length: 256<br /><br />v=3D0<br />o=3DAudiocodesGW 260=
533184 260532796 IN IP4 192.168.2.111<br />s=3DPhone-Call<br />c=3DIN IP4=
192.168.2.111<br />t=3D0 0<br />m=3Daudio 6290 RTP/AVP 18 101<br />a=3Dr=
tpmap:18 g729/8000<br />a=3Dfmtp:18 annexb=3Dno<br />a=3Drtpmap:101 telep=
hone-event/8000<br />a=3Dfmtp:101 0-15<br />a=3Dptime:20<br />a=3Dsendrec=
v<br /><br /><br /><br />----------------26:16.529----------------<br />R=
CV: XOR=3D0 560 Bytes from RCVADDR: 192.168.2.111:RCVPORT: 5060:UDP (SIP/=
2.0 486 Busy Here)<br />SIP/2.0 486 Busy Here<br />From: <sip:19=
2.168.2.48>;tag=3D9ecf126ddaf81810907bcd703c2fcbfe<br />To: sip:8726@1=
92
> .168.2.111;tag=3D1c260462125<br />Via: SIP/2.0/UDP 192.168.2.48:5060;i=
id=3D1;branch=3Dz9hG4bK9ecf126ddaf81810907acd703c2fcbfe;uas-addr=3D192.16=
8.2.111<br />CSeq: 4711 INVITE<br />Call-ID: 9ecf126d-daf8-1810-9ecc-cd70=
3c2fcbfe<br />Contact: <sip:23@192.168.2.111;user=3Dphone><br />Ser=
ver: Audiocodes-Sip-Gateway-Mediant 2000/v.4.60A.034<br />Allow: REGISTER=
, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIB=
E, UPDATE<br />Supported: em, timer, replaces, path<br />Content-Length: =
0<br /><br /><br />----------------26:16.534----------------<br />SEND: X=
OR=3D0 398 Bytes to 192.168.2.111:5060:UDP (ACK sip:8726@192.168.2.111 SI=
P/2.0) Interface Address=3D192.168.2.48<br />ACK sip:8726@192.168.2.111 S=
IP/2.0<br />From: <sip:192.168.2.48>;tag=3D9ecf126ddaf8181090=
7bcd703c2fcbfe<br />To: sip:8726@192.168.2.111;tag=3D1c260462125<br />Via=
: SIP/2.0/UDP 192.168.2.48:5060;iid=3D1;branch=3Dz9hG4bK9ecf126ddaf818109=
07acd703c2fcbfe;uas-addr=3D192.168.2.111;rport<br />CSeq: 4711 ACK<
> br />Call-ID: 9ecf126d-d<br />
>
> -----------------------------------------------------------------------=
--
> 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
>
>
> =20
|
|
From: tomach <to...@dg...> - 2007-06-08 09:13:37
|
When I tried to compile only atlsip first it gave me this kind of error:<br /><br />Error 1 fatal error C1083: Cannot open include file: 'ptbuildopts.h': No such file or <br />directory d:\repozytoria\atlrepozytorium\opensipstack\include\ptlib.h 163 <br /><br />Any suggestions?<br /> |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-08 09:25:11
|
Hi, Get OpenSIPStack project and then start a clean build. This should be the order of your build: OpenSIPStack -> ATLSIP -> Register ATLSIP -> ATLSIPSample - Ilian tomach wrote: > When I tried to compile only atlsip first it gave me this kind of error:<br /><br />Error 1 fatal error C1083: Cannot open include file: 'ptbuildopts.h': No such file or <br />directory d:\repozytoria\atlrepozytorium\opensipstack\include\ptlib.h 163 <br /><br />Any suggestions?<br /> > > ------------------------------------------------------------------------- > 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...> - 2007-06-08 10:41:14
|
Thanks for the answers!<br /><br />I did like this:<br />1. I downloaded from cvs the head revision of opensipstack and atlsip<br />2. I tried to comile with release option openSIPstack and I got this error:<br /><br />Error 1 fatal error C1083: Cannot open include file: 'B2BIVRInterface.h': No such file or directory d:\repozytoria\atlrepozytorium\opensipstack\include\B2BUA.h 152 <br /><br />I think I did everythign as you adivsed and I stil encountered this error. Whats wrong?<br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-08 11:38:17
|
tomach wrote: > Thanks for the answers!<br /><br />I did like this:<br />1. I downloaded from cvs the head revision of opensipstack and atlsip<br />2. I tried to comile with release option openSIPstack and I got this error:<br /><br />Error 1 fatal error C1083: Cannot open include file: 'B2BIVRInterface.h': No such file or directory d:\repozytoria\atlrepozytorium\opensipstack\include\B2BUA.h 152 <br /><br />I think I did everythign as you adivsed and I stil encountered this error. Whats wrong?<br /> > > Try CVS again. I have committed the missing files a while ago. -- Joegen E. Baclor Solegy LLC |
|
From: tomach <to...@dg...> - 2007-06-11 09:46:09
|
OK I downloaded new library with CVS. <br />I tried to compile ATLSIP and openstack. If I understand correct atlsip is dependened on opensipstack. So first I tried to build opensipstac (release option) and I got error:<br /><br />Error 1 error C2061: syntax error : identifier '__RPC__in_opt' d:\sdkframework3.0\include\sapi51.h 1066 <br /><br />I do not understand why this problme happends?<br />Is it microsoft problme since it belongs to include in framework3.0 directory?<br /><br /><br /><br /> |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-13 12:34:04
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Hi, It seems you have a problem with the Speech API lib and you're using SDK Framework 3.0. Is this the SDK for Windows Vista? I'm not sure if OpenSIPStack is compatible with the WinVista SDK. You can try uninstalling it though. - Ilian tomach wrote: > OK I downloaded new library with CVS. <br />I tried to compile ATLSIP and openstack. If I understand correct atlsip is dependened on opensipstack. So first I tried to build opensipstac (release option) and I got error:<br /><br />Error 1 error C2061: syntax error : identifier '__RPC__in_opt' d:\sdkframework3.0\include\sapi51.h 1066 <br /><br />I do not understand why this problme happends?<br />Is it microsoft problme since it belongs to include in framework3.0 directory?<br /><br /><br /><br /> > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: tomach <to...@dg...> - 2007-06-18 08:04:55
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I do not have windows Vista. I just have normal update to winxp sp2. .Net framework 3.0. Unfortunately I can not uninstall it right now :(.<br /> |
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From: Andre S. <eds...@ya...> - 2007-06-08 09:26:00
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Clean Solution then Build first the realease configuration tomach <to...@dg...> wrote: When I tried to compile only atlsip first it gave me this kind of error: Error 1 fatal error C1083: Cannot open include file: 'ptbuildopts.h': No such file or directory d:\repozytoria\atlrepozytorium\opensipstack\include\ptlib.h 163 Any suggestions? ------------------------------------------------------------------------- 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 --------------------------------- Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. |
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