Re: [Linphone-developers] Linphone: BYE problem

[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
> >>
> >> =============================================================================================
> >>
> >