Re: [Linphone-developers] Linphone: BYE problem

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Re: [Linphone-developers] Linphone: BYE problem

From:	Simon Morlat
Subject:	Re: [Linphone-developers] Linphone: BYE problem
Date:	Fri, 30 Apr 2010 15:01:21 +0200
Hi Joerg,

Thank you for reporting this bug.
I will fix it.
Meantime you can add --disable-libv4l to make the configure pass.

Simon

Le vendredi 30 avril 2010 à 13:21 +0200, Joerg Bergmann a écrit :
> Dear Simon,
> 
> tried your new version 3.2.99.7. I want to disable video.
> So I called
> ./configure --disable-video
> and got the following error:
> ...
> checking linux/videodev.h usability... yes
> checking linux/videodev.h presence... yes
> checking for linux/videodev.h... yes
> checking linux/videodev2.h usability... yes
> checking linux/videodev2.h presence... yes
> checking for linux/videodev2.h... yes
> checking for LIBV4L2... no
> No libv4l2 found.
> checking for LIBV4L1... no
> No libv4l1 found.
> configure: error:
> Missing libv4l2. It is highly recommended to build with
> libv4l2 headers and library. Many camera will won't work or will crash
> your application if libv4l2 is not installed.
> If you know what you are doing, you can use --disable-libv4l to disable
> 
> Why do I need that video-stuff when I want to diable video?
> Any hints?
> 
> Joerg Bergmann
> 
> Am 30.04.2010 12:48, schrieb Simon Morlat:
> > The bug you are facing here is relative to this 3.2.99.3. I discovered
> > it after advicing you to use it. Sorry.
> > Please try lastest :
> > sources:
> > http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/source/linphone-3.2.99.7.tar.gz
> > windows binary:
> > http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/win32/linphone-3.2.99.7-setup.exe
> >
> > Should work fine.
> >
> > Note: the upload of the windows file is in progress and should be
> > finished in half an our max.
> >
> > Simon
> >
> > e jeudi 29 avril 2010 à 18:02 +0200, Petr Kuba a écrit :
> >    
> >> Hi,
> >>
> >> Please find the log attached. File complete.log contains complete log,
> >> files 01.log to 05.log contain logs for individual calls.
> >>
> >> Calls 1 and 2 were connected, calls 3, 4, and 5 were not. It means that
> >> something wrong probably happened at the end of call 2.
> >>
> >> We used version 3.2.99.3 on windows.
> >>
> >> Thanks,
> >> Petr
> >>
> >> On 22.4.2010 17:22, Simon Morlat wrote:
> >>      
> >>> Hi,
> >>>
> >>> This is very strange.
> >>> COuld you please try with the new version here:
> >>> http://download.savannah.gnu.org/releases-noredirect/linphone/unstable/source/linphone-3.2.99.4.tar.gz
> >>> and if the problem still happens, please send a log collected with
> >>> linphonec -d 6
> >>> or linphone-3 --verbose
> >>>
> >>> Simon
> >>>
> >>> Le mercredi 21 avril 2010 à 12:22 +0200, Petr Kuba a écrit :
> >>>        
> >>>> Hi,
> >>>>
> >>>> We've met a problem in Linphone/3.2.0. We use command line version with
> >>>> auto-answer mode and Asterisk/1.6.1.11 as PBX.
> >>>>
> >>>> After a few calls (Linphone is a callee) where caller terminates the
> >>>> calls, the following problem occurs:
> >>>> Linphone sends OK for BYE, but Linphone call does not terminate.
> >>>> Therefore the following call is not accepted.
> >>>>
> >>>> Complete log of SIP communication with included comments is below.
> >>>>
> >>>> Thanks for help,
> >>>> Petr Kuba
> >>>>
> >>>> =============================================================================================
> >>>>
> >>>> REGISTER sip:192.168.10.50 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK29616
> >>>> From:<sip:address@hidden>;tag=18366
> >>>> To:<sip:address@hidden>
> >>>> Call-ID: 16278
> >>>> CSeq: 5 REGISTER
> >>>> Contact:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Authorization: Digest username="832", realm="asterisk",
> >>>> nonce="532b68bf", uri="sip:192.168.10.50",
> >>>> response="18734798b0b509cd4683ade8ce3d38ec", algorithm=MD5
> >>>> Max-Forwards: 70
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Expires: 900
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 401 Unauthorized
> >>>> Via: SIP/2.0/UDP
> >>>> 192.168.10.135:5060;branch=z9hG4bK29616;received=192.168.10.135;rport=5060
> >>>> From:<sip:address@hidden>;tag=18366
> >>>> To:<sip:address@hidden>;tag=as0784eb4a
> >>>> Call-ID: 16278
> >>>> CSeq: 5 REGISTER
> >>>> Server: Asterisk PBX 1.6.1.11
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> >>>> nonce="5dea0df5"
> >>>> Content-Length: 0
> >>>>
> >>>> REGISTER sip:192.168.10.50 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK16800
> >>>> From:<sip:address@hidden>;tag=18366
> >>>> To:<sip:address@hidden>
> >>>> Call-ID: 16278
> >>>> CSeq: 6 REGISTER
> >>>> Contact:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Authorization: Digest username="832", realm="asterisk",
> >>>> nonce="5dea0df5", uri="sip:192.168.10.50",
> >>>> response="1d8463cc6a9f1c030b3022181feef7fa", algorithm=MD5
> >>>> Max-Forwards: 70
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Expires: 900
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP
> >>>> 192.168.10.135:5060;branch=z9hG4bK16800;received=192.168.10.135;rport=5060
> >>>> From:<sip:address@hidden>;tag=18366
> >>>> To:<sip:address@hidden>;tag=as0784eb4a
> >>>> Call-ID: 16278
> >>>> CSeq: 6 REGISTER
> >>>> Server: Asterisk PBX 1.6.1.11
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> Expires: 900
> >>>> Contact: sip:address@hidden:5060;line=33a73e881a69e9b;expires=900
> >>>> Date: Tue, 20 Apr 2010 08:02:57 GMT
> >>>> Content-Length: 0
> >>>>
> >>>> =============================================================================================
> >>>> Incoming call is automatically aanswered by linphone. Remote party
> >>>> disconnects.
> >>>> =============================================================================================
> >>>> INVITE sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Contact:<sip:address@hidden>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> Date: Tue, 20 Apr 2010 08:09:10 GMT
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> Content-Type: application/sdp
> >>>> Content-Length: 266
> >>>>
> >>>> v=0
> >>>> o=root 1326207081 1326207081 IN IP4 192.168.10.50
> >>>> s=Asterisk PBX 1.6.1.11
> >>>> c=IN IP4 192.168.10.50
> >>>> t=0 0
> >>>> m=audio 18004 RTP/AVP 0 101
> >>>> a=rtpmap:0 PCMU/8000
> >>>> a=rtpmap:101 telephone-event/8000
> >>>> a=fmtp:101 0-16
> >>>> a=silenceSupp:off - - - -
> >>>> a=ptime:20
> >>>> a=sendrecv
> >>>> SIP/2.0 100 Trying
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 101 Dialog Establishement
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> Contact:<sip:address@hidden:5060>
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 180 Ringing
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> Contact:<sip:address@hidden:5060>
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK45b0ac87;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> Contact:<sip:address@hidden:5060>
> >>>> Content-Type: application/sdp
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length:   183
> >>>>
> >>>> v=0
> >>>> o=832 123456 654321 IN IP4 192.168.10.135
> >>>> s=A conversation
> >>>> c=IN IP4 192.168.10.135
> >>>> t=0 0
> >>>> m=audio 7078 RTP/AVP 0 101
> >>>> a=rtpmap:0 PCMU/8000
> >>>> a=rtpmap:101 telephone-event/8000
> >>>> ACK sip:address@hidden:5060 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK023625f4;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Contact:<sip:address@hidden>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 ACK
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> Content-Length: 0
> >>>>
> >>>> BYE sip:address@hidden:5060 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK61bf61a5;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Call-ID: address@hidden
> >>>> CSeq: 103 BYE
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> X-Asterisk-HangupCause: Normal Clearing
> >>>> X-Asterisk-HangupCauseCode: 16
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK61bf61a5;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1703441e
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=41
> >>>> Call-ID: address@hidden
> >>>> CSeq: 103 BYE
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> =============================================================================================
> >>>> Linphone confirms BYE but it looks like it is still connected.
> >>>> In the following call Linphone doesn't send 180 Ringing.
> >>>> The call was interrupted by Linphone user (see CANCEL below) after more
> >>>> than 20s from INVITE.
> >>>> =============================================================================================
> >>>> INVITE sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Contact:<sip:address@hidden>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> Date: Tue, 20 Apr 2010 08:15:44 GMT
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> Content-Type: application/sdp
> >>>> Content-Length: 264
> >>>>
> >>>> v=0
> >>>> o=root 729825232 729825232 IN IP4 192.168.10.50
> >>>> s=Asterisk PBX 1.6.1.11
> >>>> c=IN IP4 192.168.10.50
> >>>> t=0 0
> >>>> m=audio 19580 RTP/AVP 0 101
> >>>> a=rtpmap:0 PCMU/8000
> >>>> a=rtpmap:101 telephone-event/8000
> >>>> a=fmtp:101 0-16
> >>>> a=silenceSupp:off - - - -
> >>>> a=ptime:20
> >>>> a=sendrecv
> >>>> SIP/2.0 100 Trying
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 101 Dialog Establishement
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> Contact:<sip:address@hidden:5060>
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> CANCEL sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 CANCEL
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 CANCEL
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 487 Request Cancelled
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport=5060
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 INVITE
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Content-Length: 0
> >>>>
> >>>> ACK sip:address@hidden:5060;line=33a73e881a69e9b SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.50:5060;branch=z9hG4bK6a8182d0;rport
> >>>> Max-Forwards: 70
> >>>> From: "GTS"<sip:address@hidden>;tag=as1a191a92
> >>>> To:<sip:address@hidden:5060;line=33a73e881a69e9b>;tag=18467
> >>>> Contact:<sip:address@hidden>
> >>>> Call-ID: address@hidden
> >>>> CSeq: 102 ACK
> >>>> User-Agent: Asterisk PBX 1.6.1.11
> >>>> Content-Length: 0
> >>>>
> >>>> =============================================================================================
> >>>> REGISTER sip:192.168.10.50 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK20051
> >>>> From:<sip:address@hidden>;tag=30845
> >>>> To:<sip:address@hidden>
> >>>> Call-ID: 13720
> >>>> CSeq: 1 REGISTER
> >>>> Contact:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Max-Forwards: 70
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Expires: 900
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 401 Unauthorized
> >>>> Via: SIP/2.0/UDP
> >>>> 192.168.10.135:5060;branch=z9hG4bK20051;received=192.168.10.135;rport=5060
> >>>> From:<sip:address@hidden>;tag=30845
> >>>> To:<sip:address@hidden>;tag=as64b65fec
> >>>> Call-ID: 13720
> >>>> CSeq: 1 REGISTER
> >>>> Server: Asterisk PBX 1.6.1.11
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", 
> >>>> nonce="39dc1532"
> >>>> Content-Length: 0
> >>>>
> >>>> REGISTER sip:192.168.10.50 SIP/2.0
> >>>> Via: SIP/2.0/UDP 192.168.10.135:5060;rport;branch=z9hG4bK18733
> >>>> From:<sip:address@hidden>;tag=30845
> >>>> To:<sip:address@hidden>
> >>>> Call-ID: 13720
> >>>> CSeq: 2 REGISTER
> >>>> Contact:<sip:address@hidden:5060;line=33a73e881a69e9b>
> >>>> Authorization: Digest username="832", realm="asterisk",
> >>>> nonce="39dc1532", uri="sip:192.168.10.50",
> >>>> response="79771bbc0f50febb9ee095909ffe00aa", algorithm=MD5
> >>>> Max-Forwards: 70
> >>>> User-Agent: Linphone/3.2.0 (eXosip2/3.3.0)
> >>>> Expires: 900
> >>>> Content-Length: 0
> >>>>
> >>>> SIP/2.0 200 OK
> >>>> Via: SIP/2.0/UDP
> >>>> 192.168.10.135:5060;branch=z9hG4bK18733;received=192.168.10.135;rport=5060
> >>>> From:<sip:address@hidden>;tag=30845
> >>>> To:<sip:address@hidden>;tag=as64b65fec
> >>>> Call-ID: 13720
> >>>> CSeq: 2 REGISTER
> >>>> Server: Asterisk PBX 1.6.1.11
> >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> >>>> Supported: replaces, timer
> >>>> Expires: 900
> >>>> Contact: sip:address@hidden:5060;line=33a73e881a69e9b;expires=900
> >>>> Date: Tue, 20 Apr 2010 09:04:21 GMT
> >>>> Content-Length: 0
> >>>>
> >>>> =============================================================================================
> >>>>
> >>>>          
> >>>        
> >
> >
> >
> > _______________________________________________
> > Linphone-developers mailing list
> > address@hidden
> > http://lists.nongnu.org/mailman/listinfo/linphone-developers
> >    
> 
> 
> 
> _______________________________________________
> Linphone-developers mailing list
> address@hidden
> http://lists.nongnu.org/mailman/listinfo/linphone-developers
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[Linphone-developers] Linphone: BYE problem, Petr Kuba, 2010/04/21
- Re: [Linphone-developers] Linphone: BYE problem, Simon Morlat, 2010/04/22
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    - Re: [Linphone-developers] Linphone: BYE problem, Simon Morlat, 2010/04/30
    - Re: [Linphone-developers] Linphone: BYE problem, Joerg Bergmann, 2010/04/30
    - Re: [Linphone-developers] Linphone: BYE problem, Simon Morlat <=
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