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From: Matthew G. <fo...@op...> - 2007-06-08 03:01:42
|
I'm trying to get upper registration working and have hit a few road blocks.<br /><br />So far the register requests go through, hit the SIP server, and opensbc sees the 200 ok message, however the whole process seems to be taking too long for xten (my test softphone). By the time I get a 100 trying from opensbc, I've already sent 5 other register requests! By the time the 200 finally makes it to the client its far to late and xten appears to ignore it.<br /><br />My guess is the whole thing would work if OpenSBC actually replied faster to the requests - is there any way to see why it is taking so long to reply with a trying to the first register request? I've read all the docs on the site and can't find anything wrong with my configuration.<br /><br />Thanks!<br /><br /><br /> |
|
From: Blackeye1010 <bla...@gm...> - 2007-06-07 16:15:30
|
Sorry about that... To quick on the trigger.<br /><br />To other newbies like me:<br />open up the web admin and you can set it up there.<br /><br />Regards<br /> |
|
From: Blackeye1010 <bla...@gm...> - 2007-06-07 16:10:17
|
Is there a way to have OSBC listen only in some specified ethernet interfaces ?<br /> |
|
From: Blackeye1010 <bla...@gm...> - 2007-06-07 15:33:14
|
that link for subscription to the devel ml is giving an error :<br />This was at 16:31 GMT<br />===CUT<br /><h2>Bug in Mailman version 2.1.8</h2><h3>We're sorry, we hit a bug!</h3><p>Please inform the webmaster for this site of thisproblem. Printing of traceback and other system information has beenexplicitly inhibited, but the webmaster can find this information in theMailman error logs.</p><br />===CUT<br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-06 16:27:40
|
Robert Joly wrote: > Hi, > I just downloaded the latest OpenSipStack/OpenSBC and i get a compile > error related to the media Server changes underway. Essentially, the > OpenSBCDaemon tries to instantiate a new MediaServer using a c'tor > with no arguments but that c'tor has been removed in yesterday's > submit. here is the compiler output: > > OpenSBC.cxx: In member function 'virtual void > OpenSBCDaemon::OnStart(Tools::OSSAppConfig&)': > OpenSBC.cxx:457: error: no matching function for call to > 'MS::MediaServer::MediaServer()' > /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:112: > note: candidates are: > MS::MediaServer::MediaServer(UACORE::CallSessionManager*) > /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:84: > note: MS::MediaServer::MediaServer(const MS::MediaServer&) > make[1]: *** [obj_linux_x86_r/OpenSBC.o] Error 1 > make[1]: Leaving directory `/home/sipx/opensbc/latest/opensbc' > make: *** [optnoshared] Error 2 > > Cheers, > bob > Hi Bob, Thanks for letting me know right away. I have just committed the fix for this. Please wait several minutes for the change to propagate to anonymous CVS. -- Joegen E. Baclor CTO - Solegy LLC Email: joegen @ solegy.com Main: +1 212 801 2504 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 |
|
From: <jo...@op...> - 2007-06-06 16:22:03
|
Hi Robert, Thanks for letting me know right away. I have just committed the fix for this. Please wait several minutes for the change to propagate to anonymous CVS. Robert Joly wrote: > Hi, > I just downloaded the latest OpenSipStack/OpenSBC and i get a compile > error related to the media Server changes underway. Essentially, the > OpenSBCDaemon tries to instantiate a new MediaServer using a c'tor > with no arguments but that c'tor has been removed in yesterday's > submit. here is the compiler output: > > OpenSBC.cxx: In member function 'virtual void > OpenSBCDaemon::OnStart(Tools::OSSAppConfig&)': > OpenSBC.cxx:457: error: no matching function for call to > 'MS::MediaServer::MediaServer()' > /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:112: > note: candidates are: > MS::MediaServer::MediaServer(UACORE::CallSessionManager*) > /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:84: > note: MS::MediaServer::MediaServer(const MS::MediaServer&) > make[1]: *** [obj_linux_x86_r/OpenSBC.o] Error 1 > make[1]: Leaving directory `/home/sipx/opensbc/latest/opensbc' > make: *** [optnoshared] Error 2 > > Cheers, > bob > > -- Joegen E. Baclor Founder - opensipstack.org Email: joegen @ opensipstack.org Main: +1 212 801 2504 Manila: +63 2 747 3460 Mobile: +63 918 411 9064 OpenSIPStack MPL SIP Library http://www.opensipstack.org |
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From: Robert J. <rj...@gm...> - 2007-06-06 16:05:14
|
Hi, I just downloaded the latest OpenSipStack/OpenSBC and i get a compile error related to the media Server changes underway. Essentially, the OpenSBCDaemon tries to instantiate a new MediaServer using a c'tor with no arguments but that c'tor has been removed in yesterday's submit. here is the compiler output: OpenSBC.cxx: In member function 'virtual void OpenSBCDaemon::OnStart(Tools::OSSAppConfig&)': OpenSBC.cxx:457: error: no matching function for call to 'MS::MediaServer::MediaServer()' /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:112: note: candidates are: MS::MediaServer::MediaServer(UACORE::CallSessionManager*) /home/sipx/opensbc/latest/opensipstack/include/MediaServer.h:84: note: MS::MediaServer::MediaServer(const MS::MediaServer&) make[1]: *** [obj_linux_x86_r/OpenSBC.o] Error 1 make[1]: Leaving directory `/home/sipx/opensbc/latest/opensbc' make: *** [optnoshared] Error 2 Cheers, bob |
|
From: Joegen E. B. <jb...@so...> - 2007-06-06 15:02:34
|
Hi see my recent forum post <br /><br />http://www.opensourcesip.org:8080/jiveforums/thread.jspa?threadID=609&tstart=0<br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-06 15:02:32
|
hmmmn, this is defintely not an expected behavior. Make sure you are using the latest oss-application.conf.xml. If your problem persists after doing this, please subscribe to the developer mailing list at https://lists.sourceforge.net/lists/listinfo/opensipstack-devel and post a level 5 log.<br /><br />NOTE: please do not post the log here. It's just too big for a forum post.<br /><br /><br /><br />Mark Baker wrote:<br />I was exhibiting the exact same issue, so I built from CVS yesterday(6/4/07) but now the issue is that all calls I try to place return 403Forbidden. Calls worked before the rebuild, and the config has notchanged.<br /><br />Thanks!<br />- Mark Baker<br /><br /><br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-06 13:06:57
|
Upper Registration Upper Registration is the term used for the capability of OpenSBC to proxy Registrations towards an upstream Registrar. In most cases, SIP Networks already have an existing Registrar prior to the deployment of a Border Controller such as OpenSBC. For this reason, it would not be practical to transfer the existing SIP accounts to OpenSBC. Upper Registration is the solution to this scenario. While it is necessary that all SIP requests traverse OpenSBC for NAT continuity, Registrations MAY be allowed to be relayed towards an upstream Registrar while OpenSBC retains a copy of the AOR and masquerade on behalf of the the UA that send the registration request. Understanding Local and Remote Domain Concept in OpenSBC To understand how Upper Registration works, it is important to distinguish when OpenSBC processes a request as a local domain request and when it is treated as a remote domain request. Local Domain Requests - requests with a request-URI not resolving to one of OpenSBC's listener interface and port either via DNS host lookup, DNS SRV lookup or in raw IP Address form. Remote Domain Requests - requests that, when Resolved, points to an address and port tuple other than the ones listed as a listener addresses of OpenSBC. If REGISTER requests hits OpenSBC and OpenSBC is in either Full Mode or B2B Upper Reg Mode the following steps would be followed: 1. OpenSBC checks if the request-URI is bound to a local or a remote domain. 2. If the request is bound to a local domain ( request-URI resolves to OpenSBC listener address ), OpenSBC will then check if the to-URI resolves as local domain either in IP address from or via DNS resolution. If both criteria is true then OpenSBC will authenticate the REGISTER request against the accounts configured in the "Local Domain Accounts" section. In this case, OpenSBC takes the role of the Registrar. 3. If OpenSBC sees that the request-URI or the to-URI resolves to an external address, then it checks if an upstream registrar route is available in "Upper Registration Routes". Upper Registration Route Example Entry: [sip:*@interop.opensipstack.org] sip:interop.opensipstack.org To explain further what the above entry means ** [sip:*@interop.opensipstack.org] ** - This is a wild card match filter for the to-URI of a REGISTER hitting OpenSBC ** sip:interop.opensipstack.org ** - This is the address of the upstream registrar. CAVEAT: Take note that if the request-URI or the to-URI resolves to OpenSBC either in IP address or as a result of a DNS resolution, upper registration routes will not be checked even if you have existing entries for to To-URI. This is a common mistake that continously appears in the opensipstack-devel mailing list. 4. If a route to an upstream registrar exists, OpenSBC will change the Contact-URI of the REGISTER request to point to the address of OpenSBC before sending the request to the upstream registrar. This is called contact hijacking in OpenSBC lingo. The purpose of this is to always let calls towards the registering UA to pass through OpenSBC to assure proper NAT traversal. As far as the upstream registrar is concerned, the UA that just registered to it is OpenSBC. OpenSBC stores the original contact before it reqrites the contact address. This would enable OpenSBC to properly route requests bound to the UA in the future. A Real Life Example OpenSBC Address: 10.0.0.1:5060 Upper Registrar Address: 10.0.0.2:5060 Upper Registar FQDN: sip:interop.opensipstack.org Register sent by the UA towards OpenSBC REGISTER sip:10.0.0.1:5060 SIP/2.0 From: "Foo" <sip:fo...@in...>;tag=tag1234 To: "Foo" <sip:fo...@in...> Contact: "Foo" <sip:foo@10.0.0.3:5060> Via: SIP/UDP sip:10.0.0.3:5060;branch=branch1234 Content-Length: 0 When this request hits opensbc, OpenSBC will resolve the to-URI to 10.0.0.2:5060 and thus would treat it as remote domain request. Since there is a route entry in "Upper Registration Route" for an upstream registrar: [sip:*@interop.opensipstack.org] sip:interop.opensipstack.org OpenSBC will enable upper registration for this request and hijack the contact address. REGISTER sip:10.0.0.1:5060 SIP/2.0 From: "Foo" <sip:fo...@in...>;tag=tag1234 To: "Foo" <sip:fo...@in...> Contact: "Foo" <sip:foo@10.0.0.1:5060> Via: SIP/UDP sip:10.0.0.1:5060;branch=branch5678 Via: SIP/UDP sip:10.0.0.3:5060;branch=branch1234 Content-Length: 0 |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-06 11:50:04
|
Hi, Looks like this really is ATLSIP's fault. The password is not included in the Proxy Authorization hash. If you really need this now, a quick fix would be calling ATLSIP->InitializeSIP() prior to making a call. I'll try to come up with a better solution and check it in later. Regards, Ilian Yacine Auczone wrote: > Thanks very much Ilian, > i was able to do all that, but since the update of CNG i'm not able to > make calls, i'm always getting 403 forbidden error while i'm able to > receive calls. > Thanks > > > > > ------------------------------------------------------------------------ > > Date: Mon, 4 Jun 2007 20:40:31 +0800 > > From: ip...@so... > > To: ope...@li... > > Subject: Re: [OpenSIPStack] ATLSIP compilation steps > > > > One more thing. I think you have to rename the g729 library files to > > va_g729a.h and va_g729a.lib. > > > > - Ilian > > > > Ilian Jeri C. Pinzon wrote: > > > Hi, > > > > > > Some codecs are not enabled by default because of licensing > issues. But > > > you can enable them by downloading the codec libraries into > > > opensipstack/external/codecs/ and then recompile. > > > > > > A G.723.1 codec can be downloaded from the MyPhone project at > > > Sourceforge (http://myphone.sourceforge.net). A G.729 codec can be > > > downloaded from the VoiceAge site > > > (http://www.acelp.net/openinit_g729.php) . Take note, these codecs > are > > > bound by each of their own license restrictions so you will not be > able > > > to use them legally in certain scenarios. Use these at your own risk. > > > > > > Regards, > > > Ilian > > > > > > web...@dz... wrote: > > > > > >> Hello, > > >> > > >> Thank you very much Joegen, the problem was that i was trying to > compile it on a WIndows XP home edition computer, i succeed now on > windows xp pro, i just noticed that some codecs are not available now > whene i compiled ossphone, do i have to enable theme somewhere before > starting compilation ? > > >> > > >> thanks > > >> > > >> > > >> > > >> On Sat, 02 Jun 2007 09:49:21 +0800, "Joegen E. Baclor" > <jb...@so...> wrote: > > >> > > >> > > >> > > >>> Hi, > > >>> > > >>> > > >> > > >> > > >> > > >> > > >>> Can you copy and paste the error from the MSVC output window? > What is > > >>> > > >>> > > >> > > >> > > >>> your OS version? > > >>> > > >>> > > >> > > >> > > >> > > >> > > >>> Joegen > > >>> > > >>> > > >> > > >> > > >> > > >> > > >>> web...@dz... wrote: > > >>> > > >>> > > >> > > >> > > >>>> Hello, > > >>>> > > >>>> > > >> > > >> > > >> > > >> > > >>>> I have the latest cvs opensipstack and atlsip, i'm using microsoft > > >>>> > > >>>> > > >> > > >> > > >>> visual c++ 2005 (not express edition), i folowed the steps on how to > > >>> > > >>> > > >> > > >> > > >>> compile opensipstack from the wiki, using release build first but it > > >>> > > >>> > > >> > > >> > > >>> faild, is there any requirements before starting the compilation > process, > > >>> > > >>> > > >> > > >> > > >>> can i know what are the steps of compilation to have atlsip > compiled. > > >>> > > >>> > > >> > > >> > > >> > > >> > > >>>> Thanks for everyone. > > >>>> > > >>>> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >> > > >>> > ------------------------------------------------------------------------- > > >>> > > >>> > > >> > > >> > > >>> 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 > > >> > > >> > > >> > > > > > > > > > > ------------------------------------------------------------------------- > > > 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 > > ------------------------------------------------------------------------ > Soyez parmi les premiers à essayer Windows Live Mail. Windows Live > Mail. > <http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d> > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > 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...> - 2007-06-06 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 />
>
> -----------------------------------------------------------------------=
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>
> =20
|
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-06 09:57:25
|
Hi again. I've modified RegisterSession.cxx for this. You can now get it from the CVS. Regards, Ilian Ilian Jeri C. Pinzon wrote: > Hi Leo, > > I've reproduced this issue. Thanks for reporting this. > > Will check-in the fix. > > Regards, > Ilian > > le...@da... wrote: > >> Quoting "Joegen E. Baclor" <jb...@so...>: >> >> >> >>> Leo, >>> >>> Are you referring to the timer based refresh of the registration >>> (Registration Refresh) or are you manually calling >>> DoLogin()/SendRegister() several times? >>> >>> >> I'm doing 60s timer based refresh. I printed out the CSeq values in >> the RegisterSession method which I know is called by the refresh >> function. So far, it always looked the same. The packets that I >> capture also show that the CSeq is always the same - 2. >> >> Leo >> >> >> >>> Hi Ilian, >>> >>> Can you check Leo's findings? CSeq should be incremental. >>> >>> le...@da... wrote: >>> >>> >>>> Joegen, >>>> >>>> I noticed that the CSeq for the register message is always the same. >>>> I've tried incrementing the value manaully and calling request.SetCSeq >>>> but it's still goes back to the first value. The ConstructRegister >>>> comments says it should increment but it doesn't seem to be the case. >>>> I've tested with 1.1.16-134 and 1.1.15 >>>> >>>> Leo >>>> >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------- >>> 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 >> >> >> >> > > > ------------------------------------------------------------------------- > 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: Administrator <rn...@so...> - 2007-06-06 08:14:15
|
test <br /><br /><br />back<br /><br /><br /><br /> |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-06 06:59:59
|
Hi Leo, I've reproduced this issue. Thanks for reporting this. Will check-in the fix. Regards, Ilian le...@da... wrote: > Quoting "Joegen E. Baclor" <jb...@so...>: > > >> Leo, >> >> Are you referring to the timer based refresh of the registration >> (Registration Refresh) or are you manually calling >> DoLogin()/SendRegister() several times? >> > I'm doing 60s timer based refresh. I printed out the CSeq values in > the RegisterSession method which I know is called by the refresh > function. So far, it always looked the same. The packets that I > capture also show that the CSeq is always the same - 2. > > Leo > > >> Hi Ilian, >> >> Can you check Leo's findings? CSeq should be incremental. >> >> le...@da... wrote: >> >>> Joegen, >>> >>> I noticed that the CSeq for the register message is always the same. >>> I've tried incrementing the value manaully and calling request.SetCSeq >>> but it's still goes back to the first value. The ConstructRegister >>> comments says it should increment but it doesn't seem to be the case. >>> I've tested with 1.1.16-134 and 1.1.15 >>> >>> Leo >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by DB2 Express >> Download DB2 Express C - the FREE version of DB2 express and take >> control of your XML. No limits. Just data. Click to get it now. >> http://sourceforge.net/powerbar/db2/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: <le...@da...> - 2007-06-06 06:38:12
|
Quoting "Joegen E. Baclor" <jb...@so...>: > Leo, > > Are you referring to the timer based refresh of the registration > (Registration Refresh) or are you manually calling > DoLogin()/SendRegister() several times? I'm doing 60s timer based refresh. I printed out the CSeq values in =20 the RegisterSession method which I know is called by the refresh =20 function. So far, it always looked the same. The packets that I =20 capture also show that the CSeq is always the same - 2. Leo > > > Hi Ilian, > > Can you check Leo's findings? CSeq should be incremental. > > le...@da... wrote: >> Joegen, >> >> I noticed that the CSeq for the register message is always the same. >> I've tried incrementing the value manaully and calling request.SetCSeq >> but it's still goes back to the first value. The ConstructRegister >> comments says it should increment but it doesn't seem to be the case. >> I've tested with 1.1.16-134 and 1.1.15 >> >> Leo >> >> >> > > > ------------------------------------------------------------------------- > 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: Harley S. <fo...@op...> - 2007-06-06 04:51:33
|
I am seeing the same issue. In any OpenSBC mode it looks like it does not follow the Upper Registration Routes. OpenSBC always attempts to auth the incoming registration via it's internal database.<br /><br />Any suggestions on how I might be able to get past this issue?<br /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-06 02:16:32
|
Leo, Are you referring to the timer based refresh of the registration (Registration Refresh) or are you manually calling DoLogin()/SendRegister() several times? Hi Ilian, Can you check Leo's findings? CSeq should be incremental. le...@da... wrote: > Joegen, > > I noticed that the CSeq for the register message is always the same. > I've tried incrementing the value manaully and calling request.SetCSeq > but it's still goes back to the first value. The ConstructRegister > comments says it should increment but it doesn't seem to be the case. > I've tested with 1.1.16-134 and 1.1.15 > > Leo > > > |
|
From: <le...@da...> - 2007-06-06 00:10:22
|
Joegen, I noticed that the CSeq for the register message is always the same. I've tried incrementing the value manaully and calling request.SetCSeq but it's still goes back to the first value. The ConstructRegister comments says it should increment but it doesn't seem to be the case. I've tested with 1.1.16-134 and 1.1.15 Leo |
|
From: tomach <to...@dg...> - 2007-06-05 13:48:52
|
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 /> |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-05 11:22:19
|
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
>
>
>
|
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-05 11:19:55
|
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 > > > |
|
From: tomach <to...@dg...> - 2007-06-05 11:11:00
|
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 /> |
|
From: Joegen E. B. <jb...@so...> - 2007-06-05 10:13:34
|
Hi Everyone, I have recently uploaded a bunch of files in CVS for OpenVXI Speech browser. The library can be found at http://www.speech.cs.cmu.edu/openvxi/. This is in preparation for the much awaited IVR support in OpenSBC. OpenVXI plays the role of the VXML browser for the OpenSBC Media Server. This new capability opens new horizons where OpenSBC can be used. Some use cases would include Auto Attendants, Calling Card applications, Voice Mails etc. There are also plans of putting the MWI package into action to support Voice Mails Subscriptions in OpenSBC. Media server support is currently undergoing massive coding on my end. This may result to unsynchronized Visual Studio projects in CVS particularly (VC 7.1). Unix make environment might also lag behind since my environment is Visual Studio 2005. Great care is being observed to make sure the unix build environment is not affected by the changes in the media server codes. Please report compile errors as soon as you find them. Thank very much to all! Joegen |
|
From: Ilian J. C. P. <ip...@so...> - 2007-06-05 09:17:39
|
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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