Thread: Re: [OpenSIPStack] Problem with OnOutgoingCallConnected
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From: tomach <to...@dg...> - 2007-06-18 08:08:04
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I undefined P_SAPI as You adviced. Process of compilation went good. But now everytime I want to build softphone based on activex it does not work at all...I just try to make call and nothing happends...nothing at all....have no idae why? maybe this P_SAPI was important finally?<br /> |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-18 08:20:22
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Hi, What program did you try to run? Or do you have your own custom program that accesses ATLSIP? Can I see your logs? Regards, Ilian tomach wrote: > I undefined P_SAPI as You adviced. Process of compilation went good. But now everytime I want to build softphone based on activex it does not work at all...I just try to make call and nothing happends...nothing at all....have no idae why? maybe this P_SAPI was important finally?<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 13:57:17
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Sorry but I guess I do not understand one thing. <br />Ater I download whole CVS. I search for ptbuildopts.h. But there isnt any. There is only ptbuildsopt.h.in and in this file isnt deifned P_SAPI????<br />File ptbuildsopt.h is created during compilation so how can I undefine it? When before compilation there is no such a file?<br />I have no idea how I compiled it two days ago, but now after downloading it from cvs it is not working:( have you changed repository??<br /> |
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From: Joegen E. B. <joe...@gm...> - 2007-06-18 16:46:27
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tomach wrote: > Sorry but I guess I do not understand one thing. <br />Ater I download whole CVS. I search for ptbuildopts.h. But there isnt any. There is only ptbuildsopt.h.in and in this file isnt deifned P_SAPI????<br />File ptbuildsopt.h is created during compilation so how can I undefine it? When before compilation there is no such a file?<br />I have no idea how I compiled it two days ago, but now after downloading it from cvs it is not working:( have you changed repository??<br /> > > ptbuildopts.h is automatically generated by the build process. You can do one of two things. 1. Stop the build process just after you see that the first .cxx/.cpp file has started compiling (By this time ptbuildopts would have been generated). You can then undef P_SAPI from ptbuiltopts.h and rebuild. The problem with this approach is that every time you perform a "clean", ptbuildopts.h will be deleted and you need to start the process allover. 2. Delete the following line in "ossbuilopts.h.in" to permanently disable P_SAPI ///////////////////////////////////////////////// // // SAPI speech API (Windows only) // #undef P_SAPI #if defined(_MSC_VER) && P_SAPI #pragma include_alias(<sphelper.h>, <@SAPI_DIR@/include/sphelper.h>) #pragma include_alias(<sapi.h>, <@SAPI_DIR@/include/sapi.h>) #pragma include_alias(<sapiddk.h>, <@SAPI_DIR@/include/sapiddk.h>) #pragma include_alias(<SPError.h>, <@SAPI_DIR@/include/SPError.h>) #pragma include_alias(<SPDebug.h>, <@SAPI_DIR@/include/SPDebug.h>) #define P_SAPI_LIBRARY "@SAPI_DIR@/Lib/i386/sapi.lib" #endif |
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From: tomach <to...@dg...> - 2007-06-19 08:52:05
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Hello! 1. I would like to ask what is P_SAPI for? Even I compiled ATL I recieved many warnings. is it dangerous? 2. I would also ask for G729. I found out that I have to download G729 from http://www.acelp.net/openinit_g729.php and copied it to opensipstack/external/codecs/. I did so but when I compiled it and then run ossphone there wasnt option like g.729 :( What should i change? 3. Last thing is that when I use ossPhone I noticed that very often like in 90% when i run phone I can not change codec(it always uses gsm 6.10) even i change it on graphical interface, its still gsm 6.10(I checked it on ethereal...that rtp is all the time gsm 6.10... Please help me if you can... |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-19 09:24:06
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Hi, tomach wrote: > Hello! > > 1. I would like to ask what is P_SAPI for? Even I compiled ATL I recieved many warnings. is it dangerous? > P_SAPI is for text-to-speech capabilities. ATLSIP does not need this. Warnings like: "warning LNK4221: no public symbols found; archive member will be inaccessible?" No. You need not worry. > 2. I would also ask for G729. I found out that I have to download G729 from http://www.acelp.net/openinit_g729.php and copied it to opensipstack/external/codecs/. I did so but when I compiled it and then run ossphone there wasnt option like g.729 :( What should i change? > Try renaming the files to va_g729a.h and va_g729a.lib and put them in /opensipstack/external/codecs and then do a rebuild (release build first). > 3. Last thing is that when I use ossPhone I noticed that very often like in 90% when i run phone I can not change codec(it always uses gsm 6.10) even i change it on graphical interface, its still gsm 6.10(I checked it on ethereal...that rtp is all the time gsm 6.10... > Hmmm. I see your problem. I'll review this later. Regards, Ilian > Please help me if you can... > > ------------------------------------------------------------------------- > 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...> - 2007-06-19 10:10:48
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Ok, I am just trying everything as you said. Should I build debug version after release or release version is the most stable and totaly enough? |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-19 10:26:07
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Hi, tomach wrote: > Ok, I am just trying everything as you said. > > Should I build debug version after release or release version is the most stable and totaly enough? > Debug version is only needed when you need debug the program. So you may not need it. The release version has no debug info. You'll notice that the release version is usually smaller in size. I just told you to rebuild using release version so that the project can be configured properly (as others have pointed out in this forum). Regards, Ilian > ------------------------------------------------------------------------- > 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...> - 2007-06-19 11:51:16
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Hi! Really thansk a lot for a help!!! So till now everythign seems to work ok :). I just have to additoinal questions: 1. How often should i call initializeSIP(). What this function is doing? 2. If and how ATLSIP supports LDAP? Does it support open LDAP? |
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From: Joegen E. B. <joe...@gm...> - 2007-06-19 12:08:59
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tomach wrote: > Hi! > > Really thansk a lot for a help!!! > > So till now everythign seems to work ok :). I just have to additoinal questions: > > 1. How often should i call initializeSIP(). What this function is doing? > Every time you change any property of the softphone. > 2. If and how ATLSIP supports LDAP? Does it support open LDAP? > No. LDAP is not implemented in ATLSIP. |
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From: tomach <to...@dg...> - 2007-06-20 08:32:15
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Hello! Ok call connections works fine etc... I would like to ask how your activex works? How your activex works? When call is created? is it working in separate thread then main winform application? Bacause I noticed that winforms do not hang and react normally so its ok (main thread is not stoped). But my problem is that when I run your application (sample VB) under visual studio events come but they do NOT update controls. When I run the same application under cmd line everything works fine events come and update controls. Do you had similar reaction under Visual Studio 2005? I am really curious why is it happening? any suggestions? |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-20 10:41:58
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Hello.
tomach wrote:
> Hello!
>
> Ok call connections works fine etc...
> I would like to ask how your activex works?
> How your activex works? When call is created? is it working in separate thread then main winform application? Bacause I noticed that winforms do not hang and react normally so its ok (main thread is not stoped).
>
> But my problem is that when I run your application (sample VB) under visual studio events come but they do NOT update controls. When I run the same application under cmd line everything works fine events come and update controls.
>
> Do you had similar reaction under Visual Studio 2005?
>
Hmmm.
Yep. ATLSIP events come from a separate thread...
.NET is strict with regards to cross-thread UI updates (especially with
VS 2005). You may need to marshall UI updates to the UI thread for the
update to work.
For example with the ATLSIP_OnOutgoingCallTrying event, instead of
invoking lblStatus->Text directly from this separate thread you will
need to tell the UI thread to do it. Like this:
System::Void ATLSIP_OnOutgoingCallTrying(
System::Object^ sender,
AxATLSIPLib::_IOpenSIPStackCtlEvents_OnOutgoingCallTryingEvent^ e) {
// lblStatus->Text = L"Status: Trying"; // This does not work
SetText( lblStatus, L"Status: Trying" ); // This works.
}
delegate void SetTextCallback( Control^ control, String^ text);
void SetText( Control^ control, String^ text )
{
if ( control->InvokeRequired )
{
// We are NOT on the UI thread so we need to marshal
SetTextCallback^ settext = gcnew SetTextCallback( this,
&OSSPhone::Form1::SetText );
this->Invoke( settext, gcnew array<Object^>{ control, text } );
} else
{
// We are on the UI thread. No need to marshal
control->Text = text;
}
}
The same logic applies with all other kinds of UI update. Anyway, this
is .NET-specific stuff. You can search for .NET InvokeRequired for
further reading.
Regards,
Ilian
> I am really curious why is it happening? 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
>
>
>
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From: tomach <to...@dg...> - 2007-06-21 12:39:53
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Hello!
Thanks a lot for the answer.
Right now I can register. The problem is that after I am registered and try to make call ATLSIP do not send "Proxy-Authorization" information:
for example:
Proxy-Authorization: Digest username="107", realm="192.168.2.69", nonce="00027556963e1f5da6b506f697a3907f2f53637b", uri="sip:00586820726@192.168.2.69", response="0e2bd6b0a8cc2c19a7a4ad744d4e017b", algorithm=MD5
Authentication Scheme: Digest
Username: "107"
Realm: "192.168.2.69"
Nonce Value: "00027556963e1f5da6b506f697a3907f2f53637b"
Authentication URI: "sip:00586820726@192.168.2.69"
Digest Authentication Response: "0e2bd6b0a8cc2c19a7a4ad744d4e017b"
Algorithm: MD5
Above is the example what is missing in INVITE that ALTSIP is sending to sipServer.
Is there any chance to add it, that INVITE would be complete?
Best Regards,
Tom
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From: Ilian J. C. P. <ip...@so...> - 2007-06-22 07:10:29
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Hi, ATLSIP does reply with a Proxy-Authorization header in response to a "407 Proxy Authentication Required" message. Here are my logs from OSSPhone. SEND: enc=0 747 Bytes to 192.168.0.107:11000:UDP (INVITE sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:302@192.168.0.107 SIP/2.0 From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: sip:302@192.168.0.107 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport CSeq: 4711 INVITE Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Contact: "301" <sip:301@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 205 v=0 o=- 1182495843 1182495843 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 4 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:4 G723/8000 ----------------2:13:29.134---------------- RCV: enc=0 322 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP (SIP/2.0 100 Trying) SIP/2.0 100 Trying From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: sip:302@192.168.0.107 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107 CSeq: 4711 INVITE Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Content-Length: 0 ----------------2:13:29.216---------------- RCV: enc=0 567 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: <sip:302@192.168.0.107>;tag=b4512811fcf818109ff790ed26e13033 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport=5060;received=192.168.0.107 CSeq: 4711 INVITE Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Proxy-Authenticate: Digest realm=192.168.0.107, nonce="530e7db0dbb55accddef8327d684c2ca", opaque="c9d379cc8f16ee0ac7fa8a1993816fc7", algorithm=MD5 Content-Length: 0 ----------------2:13:29.257---------------- SEND: enc=0 543 Bytes to 192.168.0.107:11000:UDP (ACK sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107 ACK sip:302@192.168.0.107 SIP/2.0 From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: <sip:302@192.168.0.107>;tag=b4512811fcf818109ff790ed26e13033 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport CSeq: 4711 ACK Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Contact: "301" <sip:301@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------2:13:29.240---------------- RCV: enc=0 567 Bytes from RCVADDR: 192.168.0.107:RCVPORT: 11000:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: <sip:302@192.168.0.107>;tag=b4512811fcf818109ff790ed26e13033 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK5c022811fcf8181085d2df71ad1fd0c5;uas-addr=192.168.0.107;rport=5060;received=192.168.0.107 CSeq: 4711 INVITE Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Proxy-Authenticate: Digest realm=192.168.0.107, nonce="530e7db0dbb55accddef8327d684c2ca", opaque="c9d379cc8f16ee0ac7fa8a1993816fc7", algorithm=MD5 Content-Length: 0 ----------------2:13:29.310---------------- SEND: enc=0 945 Bytes to 192.168.0.107:11000:UDP (INVITE sip:302@192.168.0.107 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:302@192.168.0.107 SIP/2.0 From: 301 <sip:301@192.168.0.107>;tag=5c022811fcf8181085d3df71ad1fd0c5 To: sip:302@192.168.0.107 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=2258;branch=z9hG4bK6e7c2811fcf8181085d3df71ad1fd0c5;uas-addr=192.168.0.107;rport CSeq: 4712 INVITE Call-ID: 5c022811-fcf8-1810-9162-df71ad1fd0c5 Contact: "301" <sip:301@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Proxy-Authorization: Digest username="301", realm="192.168.0.107", nonce="530e7db0dbb55accddef8327d684c2ca", uri="sip:302@192.168.0.107", response="2f4837e25f547ca1d87b548a0c232986", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 205 v=0 o=- 1182495843 1182495843 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 4 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:4 G723/8000 tomach wrote: > Hello! > > Thanks a lot for the answer. > Right now I can register. The problem is that after I am registered and try to make call ATLSIP do not send "Proxy-Authorization" information: > for example: > Proxy-Authorization: Digest username="107", realm="192.168.2.69", nonce="00027556963e1f5da6b506f697a3907f2f53637b", uri="sip:00586820726@192.168.2.69", response="0e2bd6b0a8cc2c19a7a4ad744d4e017b", algorithm=MD5 > Authentication Scheme: Digest > Username: "107" > Realm: "192.168.2.69" > Nonce Value: "00027556963e1f5da6b506f697a3907f2f53637b" > Authentication URI: "sip:00586820726@192.168.2.69" > Digest Authentication Response: "0e2bd6b0a8cc2c19a7a4ad744d4e017b" > Algorithm: MD5 > > Above is the example what is missing in INVITE that ALTSIP is sending to sipServer. > Is there any chance to add it, that INVITE would be complete? > > Best Regards, > Tom > > ------------------------------------------------------------------------- > This SF.net email is sponsored by DB2 Express > Download DB2 Express C - the FREE version of DB2 express and take > control of your XML. No limits. Just data. Click to get it now. > http://sourceforge.net/powerbar/db2/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
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From: Andre S. <eds...@ya...> - 2007-06-22 07:51:45
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Ilian,
I am encountering clipping in the middle of the calls. what are the possible reasons? I will give you later the RTP
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From: Ilian J. C. P. <ip...@so...> - 2007-06-22 08:55:55
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Hi Andre, This is probably caused by the adaptive silence detection algorithm. Try using fixed silence detection. I've attached an email with the same problem and a possible solution. You may have to tweak the threshold value to find the sweet spot. Regards, Ilian Andre Silo wrote: > Ilian, > > I am encountering clipping in the middle of the calls. what are the possible reasons? I will give you later the RTP > > > --------------------------------- > Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. > ------------------------------------------------------------------------- > 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...> - 2007-06-26 10:20:20
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Hello! Yes you are correct opensipsource resends proxy authorization but it is uncorrect. I found an error in opensip. Scenario: Softphone's ip is 192.168.2.45, user/password 507/507 SIpserver ip is 192.168.2.69 Problem: I register wihtout any problem. But when I try to call: In INVITE message which is send by the opensourcesip, field "From" looks like this: From: 507 <sip:507@192.168.2.45>;tag=6204057789f818108168ffe5741f66c1 where You can see should be address 192.168.2.69 because otherwise in proxy authentication required is uncorrect realm (should be 2.69 and tehre is 2.45). When we look at the register request, we can see field from is correct: From: <sip:507@192.168.2.69>;tag=b94f4dd487f818108ff199e5a6d4f1ef Only in invite it is uncorrect. Have You met this probelm before? If You need any additionl logs, I am ready to help. Best Regards, Tom |
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From: tomach <to...@dg...> - 2007-06-26 10:31:22
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Instead of fields "From" above there should be: In invite it looks like this: From: 507 sip:507@192.168.2.45 ;tag=6204057789f818108168ffe5741f66c1 and for register: From: sip:507@192.168.2.69 ;tag=b94f4dd487f818108ff199e5a6d4f1ef |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-26 10:59:32
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Hi Tom, Hmmm. Can you check if you the latest versions of SoftphoneInterface.* and Softphone.*? This does not happen in my setup. My From headers are correct. Here are my logs: SEND: enc=0 531 Bytes to 70.42.73.140:5060:UDP (REGISTER sip:70.42.73.140 SIP/2.0) Interface Address= REGISTER sip:70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24b4836f04f9181094d7a2088e6bd734 To: sip:300@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1769;branch=z9hG4bK24b4836f04f9181094d8a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 1 REGISTER Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060;transport=udp> User-Agent: OpenSIPStack-1.1.6-166 Expires: 3600 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------5:46:55.280---------------- RCV: enc=0 537 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized) SIP/2.0 401 Unauthorized From: 300 <sip:300@70.42.73.140>;tag=24b4836f04f9181094d7a2088e6bd734 To: sip:300@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=3329;branch=z9hG4bK24b4836f04f9181094d8a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20 CSeq: 1 REGISTER Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734 Server: OpenSIPStack-1.1.6-132 WWW-Authenticate: Digest realm="70.42.73.140", nonce="0e73d21831c3e5cab945b3686a6f8fb3", opaque="c307ecf61b2e21296250e3a528cdf064", algorithm=MD5 Content-Length: 0 ----------------5:46:55.341---------------- SEND: enc=0 717 Bytes to 70.42.73.140:5060:UDP (REGISTER sip:70.42.73.140 SIP/2.0) Interface Address= REGISTER sip:70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=48f7836f04f9181094d8a2088e6bd734 To: sip:300@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1769;branch=z9hG4bK48f7836f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 2 REGISTER Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060;transport=udp> User-Agent: OpenSIPStack-1.1.6-166 Expires: 3600 Max-Forwards: 10 Authorization: Digest username="300", realm="70.42.73.140", nonce="0e73d21831c3e5cab945b3686a6f8fb3", uri="sip:70.42.73.140", response="b85374c0f0b2adcdc6c983c70ff61dd4", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------5:46:55.427---------------- RCV: enc=0 454 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP (SIP/2.0 200 OK) SIP/2.0 200 OK From: 300 <sip:300@70.42.73.140>;tag=48f7836f04f9181094d8a2088e6bd734 To: sip:300@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=3329;branch=z9hG4bK48f7836f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20 CSeq: 2 REGISTER Call-ID: 24b4836f-04f9-1810-8824-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060;transport=udp> Server: OpenSIPStack-1.1.6-132 Expires: 3600 Content-Length: 0 ----------------5:47:02.795---------------- SEND: enc=0 767 Bytes to 70.42.73.140:5060:UDP (INVITE sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: sip:613@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4711 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 229 v=0 o=- 1182855278 1182855278 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 8 0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 ----------------5:47:02.841---------------- SEND: enc=0 767 Bytes to 70.42.73.140:5060:UDP (INVITE sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: sip:613@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4711 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 229 v=0 o=- 1182855278 1182855278 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 8 0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 ----------------5:47:02.928---------------- RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20 CSeq: 4711 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Proxy-Authenticate: Digest realm=70.42.73.140, nonce="ec52dfdc48854008a4e1f5b327b6a7ab", opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5 Content-Length: 0 ----------------5:47:02.979---------------- SEND: enc=0 963 Bytes to 70.42.73.140:5060:UDP (INVITE sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: sip:613@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bKe69f8f6f04f9181094daa2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4712 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Proxy-Authorization: Digest username="300", realm="70.42.73.140", nonce="ec52dfdc48854008a4e1f5b327b6a7ab", uri="sip:613@70.42.73.140", response="3606c0c2a3553f1be2de540d929d0965", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 229 v=0 o=- 1182855278 1182855278 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 8 0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 ----------------5:47:02.944---------------- RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20 CSeq: 4711 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Proxy-Authenticate: Digest realm=70.42.73.140, nonce="ec52dfdc48854008a4e1f5b327b6a7ab", opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5 Content-Length: 0 ----------------5:47:03.040---------------- SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 ACK sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4711 ACK Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------5:47:03.029---------------- RCV: enc=0 563 Bytes from RCVADDR: 70.42.73.140:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required) SIP/2.0 407 Proxy Authentication Required From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport=5060;received=64.243.115.20 CSeq: 4711 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Proxy-Authenticate: Digest realm=70.42.73.140, nonce="ec52dfdc48854008a4e1f5b327b6a7ab", opaque="4c7f21d239815332ea078702ccd86d83", algorithm=MD5 Content-Length: 0 ----------------5:47:03.086---------------- SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 ACK sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4711 ACK Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------5:47:03.129---------------- SEND: enc=0 539 Bytes to 70.42.73.140:5060:UDP (ACK sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 ACK sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: <sip:613@70.42.73.140>;tag=e9128a6f04f918108bdfd184264b0e8e Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bK24388f6f04f9181094d9a2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4711 ACK Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Length: 0 ----------------5:47:03.337---------------- SEND: enc=0 963 Bytes to 70.42.73.140:5060:UDP (INVITE sip:613@70.42.73.140 SIP/2.0) Interface Address=192.168.0.107 INVITE sip:613@70.42.73.140 SIP/2.0 From: 300 <sip:300@70.42.73.140>;tag=24388f6f04f9181094daa2088e6bd734 To: sip:613@70.42.73.140 Via: SIP/2.0/UDP 192.168.0.107:5060;iid=1770;branch=z9hG4bKe69f8f6f04f9181094daa2088e6bd734;uas-addr=70.42.73.140;rport CSeq: 4712 INVITE Call-ID: 24388f6f-04f9-1810-8825-a2088e6bd734 Contact: "300" <sip:300@192.168.0.107:5060> User-Agent: OpenSIPStack-1.1.6-166 Max-Forwards: 10 Proxy-Authorization: Digest username="300", realm="70.42.73.140", nonce="ec52dfdc48854008a4e1f5b327b6a7ab", uri="sip:613@70.42.73.140", response="3606c0c2a3553f1be2de540d929d0965", algorithm=MD5 Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS Content-Type: application/sdp Content-Length: 229 v=0 o=- 1182855278 1182855278 IN IP4 192.168.0.107 s=OSS RTP Session c=IN IP4 192.168.0.107 t=0 0 m=audio 5000 RTP/AVP 101 8 0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 tomach wrote: > Instead of fields "From" above there should be: > > In invite it looks like this: > > From: 507 sip:507@192.168.2.45 ;tag=6204057789f818108168ffe5741f66c1 > > and for register: > > From: sip:507@192.168.2.69 ;tag=b94f4dd487f818108ff199e5a6d4f1ef > > ------------------------------------------------------------------------- > 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...> - 2007-06-26 11:49:54
|
Hi! Files that you asked and dates of their creation: Softphone.h 2007.05.22 Softphone.cxx 2007.06.04 SoftphoneInterface.h 2007.05.31 SoftphoneInterface.cxx 2007.05.30 I belive that those are the newest one. |
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From: Ilian J. C. P. <ip...@so...> - 2007-06-26 12:34:42
|
Hi, I've seen your logs and the From tag was definitely wrong but I can't reproduce this problem. By the way, what app are you using? OSSPhone? Have you by any chance coded something that sets the wrong domain before calling? Regards, Ilian tomach wrote: > Hi! > > Files that you asked and dates of their creation: > > Softphone.h 2007.05.22 > Softphone.cxx 2007.06.04 > SoftphoneInterface.h 2007.05.31 > SoftphoneInterface.cxx 2007.05.30 > > > > > I belive that those are the newest one. > > ------------------------------------------------------------------------- > 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...> - 2007-06-26 13:05:33
|
I was using both sample app and ossphone. I didnt code anything jsut run sample and try to make call. then i exmaine that everytime for FROM it took hostIP and it shoudl take domain and not my host ip..... Im trying to find where to change it but i guess it will take time....Its really weird that you can not simulate this situation.... As i said even in SampleATL i just write sip:8726@192.168.2.69 it sends invite where FROM is sip:192.168.2.45 where is should be 2.69.... weird that it only happend in my sample... |
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From: tomach <to...@dg...> - 2007-06-26 13:11:51
|
I was using both sample app and ossphone. I didnt code anything jsut run sample and try to make call. then i exmaine that everytime for FROM it took hostIP and it shoudl take domain and not my host ip..... Im trying to find where to change it but i guess it will take time....Its really weird that you can not simulate this situation.... As i said even in SampleATL i just write sip:8726@192.168.2.69 it sends invite where FROM is sip:192.168.2.45 where is should be 2.69.... weird that it only happend in my sample... |
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From: Joegen E. B. <joe...@gm...> - 2007-06-26 12:49:27
|
I just committed a patch that might solve your problem. Make a fresh CVS update. It is also important that you tell us if you made code modifications. tomach wrote: > Hi! > > Files that you asked and dates of their creation: > > Softphone.h 2007.05.22 > Softphone.cxx 2007.06.04 > SoftphoneInterface.h 2007.05.31 > SoftphoneInterface.cxx 2007.05.30 > > > > > I belive that those are the newest one. > |
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From: tomach <to...@dg...> - 2007-06-26 13:42:07
|
OK. I tried it with three different sipServers (different vendors) everytime ATLSample behave the same. FROM field was not correct :( It is really weird that it works correct with your sipServer. Best Regards, Tom |
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