Re: [Linphone-developers] Linphone: BYE problem

linphone-developers
[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
Re: [Linphone-developers] Linphone: BYE problem

From:	Simon Morlat
Subject:	Re: [Linphone-developers] Linphone: BYE problem
Date:	Thu, 22 Apr 2010 17:22:42 +0200
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
> 
> =============================================================================================
>
[Prev in Thread]
Current Thread
[Next in Thread]
[Linphone-developers] Linphone: BYE problem, Petr Kuba, 2010/04/21
- Re: [Linphone-developers] Linphone: BYE problem, Simon Morlat <=
  - Re: [Linphone-developers] Linphone: BYE problem, Petr Kuba, 2010/04/29
  - Message not available
    - 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, 2010/04/30
Prev by Date: RE: [Linphone-developers] Error in msx264 plugin?
Next by Date: Re: Re : [Linphone-developers] support SIP MESSAGE during call
Previous by thread: [Linphone-developers] Linphone: BYE problem
Next by thread: Re: [Linphone-developers] Linphone: BYE problem
Index(es):
- Date
- Thread