Re: [Linphone-users] Linphone freezes GUI on call, misses Hangup

[Juergen Sauer]

Hi Gautier,

Am 04.01.2016 um 10:09 schrieb Gautier Pelloux-Prayer:
> Could you get logs and send us them please? We get some reports from time to 
> time where application completely freezes but without logs we cannot do much. 
> Please see 
> https://wiki.linphone.org/wiki/index.php/Faq#I_have_a_problem._How_to_get_logs.2Ftools.2Fcontacts_to_troubleshoot_the_issue.3F

Ofcourse :)


So, I started
address@hidden ~]$ linphone --verbose &>linphone.error


After registration, I called our internel test number "100", timeservice
@asterisk (192.168.11.251) Console Log  off asterisk:

Connected to Asterisk 11.13.1~dfsg-2+b1 currently running on gw (pid =
12721)
Core debug is still 5.
    -- Unregistered SIP 'pc7'
    -- Registered SIP 'pc7' at 192.168.11.16:5060
       > Saved useragent "Linphone/3.9.1 (belle-sip/1.4.2)" for peer pc7
  == Using SIP RTP CoS mark 5
    -- Executing address@hidden:1] Answer("SIP/pc7-00000008", "") in new
stack
       > 0x7fd1bc0314a0 -- Probation passed - setting RTP source address
to 192.168.11.16:7078
    -- Executing address@hidden:2] Wait("SIP/pc7-00000008", "1") in new stack
    -- Executing address@hidden:3] Set("SIP/pc7-00000008",
"FUTURETIME=1451903550") in new stack
    -- Executing address@hidden:4] Set("SIP/pc7-00000008",
"TIME=1451907150") in new stack
    -- Executing address@hidden:5] SayUnixTime("SIP/pc7-00000008",
"1451907150, Europe/Berlin, HM ABdY") in new stack
    -- <SIP/pc7-00000008> Playing 'digits/11.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/oclock.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/2-and.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/30.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/day-1.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/mon-0.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/h-4.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/2.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/thousand.gsm' (language 'de')
    -- <SIP/pc7-00000008> Playing 'digits/16.gsm' (language 'de')
    -- Executing address@hidden:6] WaitUntil("SIP/pc7-00000008",
"1451903550") in new stack
    -- Executing address@hidden:7] Playback("SIP/pc7-00000008", "beep")
in new stack
    -- <SIP/pc7-00000008> Playing 'beep.gsm' (language 'de')
    -- Executing address@hidden:8] Hangup("SIP/pc7-00000008", "") in new
stack
  == Spawn extension (internal, 100, 8) exited non-zero on
'SIP/pc7-00000008'
gw*CLI>

############################################################################
linphone.error log is:

linphone-message : Using (r/w) config information from
/home/jojo/.linphonerc
linphone-message : Initializing LinphoneCore 3.9.1
linphone-message : Vtable [0x1876220] registered on core [0x1871b50]
linphone-message : Linphone core [0x1876220] notifying
[global_state_changed]
linphone-message : oRTP-0.25.0 initialized.
linphone-message : Mediastreamer2 factory 2.12.1 (git: 2.12.1) initialized.
linphone-message : CPU count set to 8
linphone-message : ms_factory_init() done: platform_tags=linux,x86,desktop
linphone-message : srtp init
linphone-message : Registering all soundcard handlers
linphone-message : New PulseAudio context state: PA_CONTEXT_CONNECTING
linphone-message : New PulseAudio context state: PA_CONTEXT_AUTHORIZING
linphone-message : New PulseAudio context state: PA_CONTEXT_SETTING_NAME
linphone-message : New PulseAudio context state: PA_CONTEXT_READY
linphone-message : Card 'PulseAudio: Internes Audio Digital Stereo
(HDMI)' added
linphone-message : Card 'PulseAudio: ClearChat Pro USB Analog Stereo' added
linphone-message : Card 'PulseAudio: Internes Audio Analog Stereo' added
linphone-message : Card 'PulseAudio: AK5370 I/F A/D Converter Analog
Mono' added
linphone-message : Card 'PulseAudio: ClearChat Pro USB Analog Mono' added
linphone-message : Card 'ALSA: default device' added
linphone-message : also error in pcm_hw.c:1590 - open
'/dev/snd/pcmC0D0c' failed (-2)
linphone-message : also error in pcm_dsnoop.c:606 - unable to open slave
linphone-message : also error in pcm_hw.c:1590 - open
'/dev/snd/pcmC0D0p' failed (-2)
linphone-message : also error in pcm_dmix.c:1029 - unable to open slave
linphone-message : Registering all webcam handlers
linphone-message : Webcam V4L2: /dev/video0 added
linphone-message : Webcam StaticImage: Static picture added
linphone-message : ms_factory_init_voip() done
linphone-message : Loading ms plugins from [/usr/lib/mediastreamer/plugins]
linphone-message : Loading plugin
/usr/lib/mediastreamer/plugins/libmsbcg729.so.0...
linphone-message :  libmsbcg729 debug plugin loaded
linphone-message : Plugin loaded
(/usr/lib/mediastreamer/plugins/libmsbcg729.so.0)
linphone-message : Codec opus/48000 fmtp=[useinbandfec=1] number=-1,
enabled=1) added to default capabilities.
linphone-message : Could not find encoder for SILK
linphone-message : Could not find decoder for SILK
linphone-message : Codec speex/16000 fmtp=[vbr=on] number=-1, enabled=1)
added to default capabilities.
linphone-message : Codec speex/8000 fmtp=[vbr=on] number=-1, enabled=1)
added to default capabilities.
linphone-message : Codec PCMU/8000 fmtp=[] number=0, enabled=1) added to
default capabilities.
linphone-message : Codec PCMA/8000 fmtp=[] number=8, enabled=1) added to
default capabilities.
linphone-message : Codec t140/1000 fmtp=[] number=96, enabled=1) added
to default capabilities.
linphone-message : Codec red/1000 fmtp=[] number=97, enabled=1) added to
default capabilities.
linphone-message : Codec GSM/8000 fmtp=[] number=3, enabled=0) added to
default capabilities.
linphone-message : Codec G722/8000 fmtp=[] number=9, enabled=0) added to
default capabilities.
linphone-message : Could not find encoder for iLBC
linphone-message : Could not find decoder for iLBC
linphone-message : Could not find encoder for AMR
linphone-message : Could not find decoder for AMR
linphone-message : Could not find encoder for AMR-WB
linphone-message : Could not find decoder for AMR-WB
linphone-message : Codec G729/8000 fmtp=[annexb=no] number=18,
enabled=0) added to default capabilities.
linphone-message : Could not find encoder for mpeg4-generic
linphone-message : Could not find decoder for mpeg4-generic
linphone-message : Could not find encoder for mpeg4-generic
linphone-message : Could not find decoder for mpeg4-generic
linphone-message : Could not find encoder for mpeg4-generic
linphone-message : Could not find decoder for mpeg4-generic
linphone-message : Could not find encoder for mpeg4-generic
linphone-message : Could not find decoder for mpeg4-generic
linphone-message : Could not find encoder for mpeg4-generic
linphone-message : Could not find decoder for mpeg4-generic
linphone-message : Could not find encoder for iSAC
linphone-message : Could not find decoder for iSAC
linphone-message : Codec speex/32000 fmtp=[vbr=on] number=-1, enabled=0)
added to default capabilities.
linphone-message : Could not find encoder for SILK
linphone-message : Could not find decoder for SILK
linphone-message : Could not find encoder for SILK
linphone-message : Could not find decoder for SILK
linphone-message : Could not find encoder for SILK
linphone-message : Could not find decoder for SILK
linphone-message : Could not find encoder for G726-16
linphone-message : Could not find decoder for G726-16
linphone-message : Could not find encoder for G726-24
linphone-message : Could not find decoder for G726-24
linphone-message : Could not find encoder for G726-32
linphone-message : Could not find decoder for G726-32
linphone-message : Could not find encoder for G726-40
linphone-message : Could not find decoder for G726-40
linphone-message : Could not find encoder for AAL2-G726-16
linphone-message : Could not find decoder for AAL2-G726-16
linphone-message : Could not find encoder for AAL2-G726-24
linphone-message : Could not find decoder for AAL2-G726-24
linphone-message : Could not find encoder for AAL2-G726-32
linphone-message : Could not find decoder for AAL2-G726-32
linphone-message : Could not find encoder for AAL2-G726-40
linphone-message : Could not find decoder for AAL2-G726-40
linphone-message : Could not find encoder for CODEC2
linphone-message : Could not find decoder for CODEC2
linphone-message : Codec VP8/90000 fmtp=[] number=-1, enabled=1) added
to default capabilities.
linphone-message : Could not find encoder for H264
linphone-message : Codec MP4V-ES/90000 fmtp=[profile-level-id=3]
number=-1, enabled=1) added to default capabilities.
linphone-message : Codec H263-1998/90000 fmtp=[CIF=1;QCIF=1] number=-1,
enabled=0) added to default capabilities.
linphone-message : Codec H263/90000 fmtp=[] number=34, enabled=0) added
to default capabilities.
linphone-message : Could not find encoder for 1016
linphone-message : Could not find decoder for 1016
linphone-message : Could not find encoder for G723
linphone-message : Could not find decoder for G723
linphone-message : Could not find encoder for LPC
linphone-message : Could not find decoder for LPC
linphone-message : Codec L16/44100 fmtp=[] number=10, enabled=0) added
to default capabilities.
linphone-message : Codec L16/44100 fmtp=[] number=11, enabled=0) added
to default capabilities.
linphone-message : Could not find encoder for CN
linphone-message : Could not find decoder for CN
linphone-message : Could not find encoder for H261
linphone-message : Could not find decoder for H261
linphone-message : Could not find encoder for MPV
linphone-message : Could not find decoder for MPV
linphone-message : Sal nat helper [enabled]
linphone-message : Root ca path set to /etc/ssl/certs
linphone-message : Root ca path set to /etc/ssl/certs
linphone-message : Root ca path set to /etc/ssl/certs
linphone-message : Linphone core [0x1876220] notifying [configuring_status]
linphone-message : Cannot open directory /usr/lib/liblinphone/plugins:
Datei oder Verzeichnis nicht gefunden
linphone-warning : no card with id PulseAudio: Logitech USB Headset
Analog Stereo
linphone-warning : no card with id PulseAudio: Logitech USB Headset
Analog Stereo
linphone-warning : no card with id PulseAudio: Logitech USB Headset
Analog Mono
linphone-message : linphone_core_set_playback_gain_db(): no active call.
linphone-message : linphone_core_set_mic_gain_db(): no active call.
linphone-message : MTU is supposed to be 1300, rtp payload max size will
be 1240
linphone-message : Sal nat helper [enabled]
linphone-message : Sal use rport [enabled]
linphone-message : Supported codec t140/1000 fmtp= automatically added
to codec list.
linphone-message : Supported codec red/1000 fmtp= automatically added to
codec list.
linphone-message : Sal use rport [enabled]
linphone-message : Root ca path set to /etc/ssl/certs
linphone-message : sal_unlisten_ports done
linphone-message : Creating listening point [0x18c55d0] on
[sip:0.0.0.0:5060;transport=UDP]
linphone-message : Creating listening point [0x18c5ae0] on
[sip:0.0.0.0:5060;transport=TCP]
linphone-message : Linphone core [0x1876220] notifying [display_status]
linphone-message : Notifying all friends that we are [online]
linphone-message : StatusIcon: Initialising
linphone-message : StatusIcon: looking for implementation...
linphone-message : Linphone core [0x1876220] notifying
[global_state_changed]
linphone-message : Table already up to date: duplicate column name: url.
linphone-message : Table already up to date: duplicate column name: utc.
linphone-message : Table already up to date: duplicate column name: appdata.
linphone-message : Table already up to date: duplicate column name: content.
linphone-message : Table already up to date: duplicate column name: call_id.
linphone-message : linphone_core_get_call_history(): completed in 2 ms
linphone-warning : nothing to migrate, skipping...
linphone-message : linphone_core_get_call_history(): completed in 3 ms
linphone-message : StatusIcon: found implementation: status_notifier
linphone-message : StatusIcon: instanciating singleton
linphone-message : StatusIcon: starting status icon
linphone-message : New local ip address is 192.168.11.16
linphone-message : Network state is now [UP]
linphone-message : LinphoneProxyConfig [0x18c5450] about to register
(LinphoneCore version: 3.9.1)
linphone-message : belle_sip_client_transaction_send_request(): waiting
channel to be ready
linphone-message : channel [0x1a54400]: starting resolution of
192.168.X.GWXX
linphone-message : channel 0x1a54400: state RES_IN_PROGRESS
linphone-message : transaction [0x1aa07d0] channel state changed to
[RES_IN_PROGRESS]
linphone-message : channel 0x1a54400: state RES_DONE
linphone-message : transaction [0x1aa07d0] channel state changed to
[RES_DONE]
linphone-message : channel 0x1a54400: state CONNECTING
linphone-message : transaction [0x1aa07d0] channel state changed to
[CONNECTING]
linphone-message : Trying to connect to [UDP://192.168.X.GWXX:5060]
linphone-message : belle_sip_get_src_addr_for(): af_inet6=0
linphone-message : Channel has local address 192.168.11.16:5060
linphone-message : channel 0x1a54400: state READY
linphone-message : transaction [0x1aa07d0] channel state changed to [READY]
linphone-message : Changing [client] [REGISTER] transaction [0x1aa07d0],
from state [INIT] to [TRYING]
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [510] bytes
REGISTER sip:192.168.X.GWXX SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.Ce6oshiHm;rport
From: <sip:address@hidden>;tag=Ra~AZ9KZQ
To: sip:address@hidden
CSeq: 20 REGISTER
Call-ID: SbYDdGdElI
Max-Forwards: 70
Supported: outbound
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact:
<sip:address@hidden>;+sip.instance="<urn:uuid:fd3a62fb-5ccf-44ef-ba6c-8886c45adf2e>"
Expires: 3600
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)


linphone-message : Neither Expires header nor corresponding Contact
header found, checking from original request
linphone-message : Refresher [0x1a9ca50] takes ownership of transaction
[0x1aa07d0]
linphone-message : Linphone core [0x1876220] notifying [display_status]
linphone-message : Proxy config [0x18c5450] for identity
[sip:address@hidden moving from state [LinphoneRegistrationNone] to
[LinphoneRegistrationProgress]
linphone-message : Linphone core [0x1876220] notifying
[registration_state_changed]
linphone-message : channel [0x1a54400]: received [502] new bytes from
[UDP://192.168.X.GWXX:5060]:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.11.16:5060;branch=z9hG4bK.Ce6oshiHm;received=192.168.11.16;rport=5060
From: <sip:address@hidden>;tag=Ra~AZ9KZQ
To: sip:address@hidden;tag=as2ac2f1d1
Call-ID: SbYDdGdElI
CSeq: 20 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="gw", nonce="0893ccfe"
Content-Length: 0


linphone-message : channel [0x1a54400] [502] bytes parsed
linphone-message : channel [0x1a54400]: discovered public ip and port
are [192.168.11.16:5060]
linphone-message : Found transaction matching response.
linphone-message : Changing [client] [REGISTER] transaction [0x1aa07d0],
from state [TRYING] to [COMPLETED]
linphone-message : linphone_core_find_auth_info(): returning auth info
username=pc7, realm=gw
linphone-message : Auth info found for [pc7] realm [gw]
linphone-message : Changing [client] [REGISTER] transaction [0x1aa2540],
from state [INIT] to [TRYING]
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [666] bytes
REGISTER sip:192.168.X.GWXX SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.ep7xs1sju;rport
From: <sip:address@hidden>;tag=Ra~AZ9KZQ
To: sip:address@hidden
CSeq: 21 REGISTER
Call-ID: SbYDdGdElI
Max-Forwards: 70
Supported: outbound
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact:
<sip:address@hidden>;+sip.instance="<urn:uuid:fd3a62fb-5ccf-44ef-ba6c-8886c45adf2e>"
Expires: 3600
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)
Authorization:  Digest realm="gw", nonce="0893ccfe", algorithm=MD5,
username="pc7",  uri="sip:192.168.X.GWXX",
response="43eac99bb9663c7f5b4cef9468752e04"


linphone-message : channel [0x1a54400]: received [559] new bytes from
[UDP://192.168.X.GWXX:5060]:
OPTIONS sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.X.GWXX:5060;branch=z9hG4bK0fc032b7
Max-Forwards: 70
From: "asterisk" <sip:address@hidden>;tag=as7a5bfc60
To: <sip:address@hidden>
Contact: <sip:address@hidden:5060>
Call-ID: address@hidden:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1
Date: Mon, 04 Jan 2016 10:32:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


linphone-message : channel [0x1a54400] [559] bytes parsed
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [263] bytes
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.X.GWXX:5060;branch=z9hG4bK0fc032b7
From: "asterisk" <sip:address@hidden>;tag=as7a5bfc60
To: <sip:address@hidden>;tag=F1rb9
Call-ID: address@hidden:5060
CSeq: 102 OPTIONS


linphone-message : channel [0x1a54400]: received [521] new bytes from
[UDP://192.168.X.GWXX:5060]:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.11.16:5060;branch=z9hG4bK.ep7xs1sju;received=192.168.11.16;rport=5060
From: <sip:address@hidden>;tag=Ra~AZ9KZQ
To: sip:address@hidden;tag=as2ac2f1d1
Call-ID: SbYDdGdElI
CSeq: 21 REGISTER
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:address@hidden>;expires=3600
Date: Mon, 04 Jan 2016 10:32:12 GMT
Content-Length: 0


linphone-message : channel [0x1a54400] [521] bytes parsed
linphone-message : Found transaction matching response.
linphone-message : Changing [client] [REGISTER] transaction [0x1aa2540],
from state [TRYING] to [COMPLETED]
linphone-message : Refresher [0x1a9ca50]:  has no contact for request
[0x18c8500].
linphone-message : Refresher: scheduling next timer in 3240000 ms
linphone-message : Register refresher  [200] reason [OK] for proxy
[sip:192.168.X.GWXX]
linphone-message : Proxy config [0x18c5450] for identity
[sip:address@hidden moving from state
[LinphoneRegistrationProgress] to [LinphoneRegistrationOk]
linphone-message : Linphone core [0x1876220] notifying
[registration_state_changed]
linphone-message : Linphone core [0x1876220] notifying [display_status]
linphone-message : Changing [client] [REGISTER] transaction [0x1aa07d0],
from state [COMPLETED] to [TERMINATED]
linphone-message : Client internal REGISTER transaction [0x1aa07d0]
terminated
linphone-message : Changing [client] [REGISTER] transaction [0x1aa2540],
from state [COMPLETED] to [TERMINATED]
linphone-message : Client internal REGISTER transaction [0x1aa2540]
terminated
linphone-message : New LinphoneCall [0x1afde90] initialized
(LinphoneCore version: 3.9.1)
linphone-message : Call 0x1afde90: moving from state LinphoneCallIdle to
LinphoneCallOutgoingInit
linphone-message : Call 0x1afde90 is locking sound resources.
linphone-message : Linphone core [0x1876220] notifying [call_state_changed]
linphone-message : Cannot determine multicast role for stream type
[audio] on call [0x1afde90]
linphone-message : RtpSession bound to [0.0.0.0] ports [7078] [7079]
linphone-message : Setting DSCP to 46 for MSAudio stream.
linphone-message : Equalizer location: hp
linphone-message : cannot set noise gate mode to [0] because no volume send
linphone-message : Cannot determine multicast role for stream type
[video] on call [0x1afde90]
linphone-message : RtpSession bound to [0.0.0.0] ports [9078] [9079]
linphone-message : Setting DSCP to 0 for MSVideo stream.
linphone-message : Contact has been fixed using proxy
linphone-message : Don't put video stream on local offer for call
[0x1afde90]
linphone-message : Don't put text stream on local offer for call [0x1afde90]
linphone-message : ms_filter_link:
MSRtpRecv:0x1af0ed0,0-->MSVoidSink:0x1ae9510,0
linphone-message : [sip:address@hidden calling
[sip:address@hidden on op [0x1afc300]
linphone-message : Skipping top route of initial route-set because same
as request-uri.
linphone-message : Changing [client] [INVITE] transaction [0x1a6bb20],
from state [INIT] to [CALLING]
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [920] bytes
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.jY1rWOTaq;rport
From: <sip:address@hidden>;tag=lZR0H0cMu
To: sip:address@hidden
CSeq: 20 INVITE
Call-ID: ugGHDKr058
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 373
Contact:
<sip:address@hidden>;+sip.instance="<urn:uuid:fd3a62fb-5ccf-44ef-ba6c-8886c45adf2e>"
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)

v=0
o=pc7 843 1926 IN IP4 192.168.11.16
s=Talk
c=IN IP4 192.168.11.16
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 18 9 97 3 101 98
a=rtpmap:96 speex/8000
a=fmtp:96 vbr=on
a=fmtp:18 annexb=no
a=rtpmap:97 speex/32000
a=fmtp:97 vbr=on
a=rtpmap:101 telephone-event/8000
a=rtpmap:98 telephone-event/32000

linphone-message : Linphone core [0x1876220] notifying [display_status]
linphone-message : Call 0x1afde90: moving from state
LinphoneCallOutgoingInit to LinphoneCallOutgoingProgress
linphone-message : Call 0x1afde90 is locking sound resources.
linphone-message : Linphone core [0x1876220] notifying [call_state_changed]
linphone-message : Priority used: 99
linphone-message : MSAudio MSTicker priority set to SCHED_RR and value (99)
linphone-message : channel [0x1a54400]: received [500] new bytes from
[UDP://192.168.X.GWXX:5060]:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.11.16:5060;branch=z9hG4bK.jY1rWOTaq;received=192.168.11.16;rport=5060
From: <sip:address@hidden>;tag=lZR0H0cMu
To: sip:address@hidden;tag=as0a014d94
Call-ID: ugGHDKr058
CSeq: 20 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="gw", nonce="7c80bc5f"
Content-Length: 0


linphone-message : channel [0x1a54400] [500] bytes parsed
linphone-message : Found transaction matching response.
linphone-message : Changing [client] [INVITE] transaction [0x1a6bb20],
from state [CALLING] to [PROCEEDING]
linphone-message : Changing [client] [INVITE] transaction [0x1a6bb20],
from state [PROCEEDING] to [COMPLETED]
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [348] bytes
ACK sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.jY1rWOTaq;rport
Call-ID: ugGHDKr058
From: <sip:address@hidden>;tag=lZR0H0cMu
To: <sip:address@hidden>;tag=as0a014d94
Contact:
<sip:address@hidden>;+sip.instance="<urn:uuid:fd3a62fb-5ccf-44ef-ba6c-8886c45adf2e>"
Max-Forwards: 70
CSeq: 20 ACK


linphone-message : linphone_core_find_auth_info(): returning auth info
username=pc7, realm=gw
linphone-message : Auth info found for [pc7] realm [gw]
linphone-message : Changing [client] [INVITE] transaction [0x1b4d3f0],
from state [INIT] to [CALLING]
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [1080] bytes
INVITE sip:address@hidden SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.rXMlLVHoU;rport
From: <sip:address@hidden>;tag=lZR0H0cMu
To: sip:address@hidden
CSeq: 21 INVITE
Call-ID: ugGHDKr058
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 373
Contact:
<sip:address@hidden>;+sip.instance="<urn:uuid:fd3a62fb-5ccf-44ef-ba6c-8886c45adf2e>"
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)
Authorization:  Digest realm="gw", nonce="7c80bc5f", algorithm=MD5,
username="pc7",  uri="sip:address@hidden",
response="73e29b9c7321f940641cc951a5bc0121"

v=0
o=pc7 843 1926 IN IP4 192.168.11.16
s=Talk
c=IN IP4 192.168.11.16
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 18 9 97 3 101 98
a=rtpmap:96 speex/8000
a=fmtp:96 vbr=on
a=fmtp:18 annexb=no
a=rtpmap:97 speex/32000
a=fmtp:97 vbr=on
a=rtpmap:101 telephone-event/8000
a=rtpmap:98 telephone-event/32000

linphone-message : channel [0x1a54400]: received [449] new bytes from
[UDP://192.168.X.GWXX:5060]:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.11.16:5060;branch=z9hG4bK.rXMlLVHoU;received=192.168.11.16;rport=5060
From: <sip:address@hidden>;tag=lZR0H0cMu
To: sip:address@hidden
Call-ID: ugGHDKr058
CSeq: 21 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:address@hidden:5060>
Content-Length: 0


linphone-message : channel [0x1a54400] [449] bytes parsed
linphone-message : Found transaction matching response.
linphone-message : Changing [client] [INVITE] transaction [0x1b4d3f0],
from state [CALLING] to [PROCEEDING]
linphone-message : op [0x1afc300] : set_or_update_dialog()
current=[(nil)] new=[(nil)]
linphone-message : Op [0x1afc300] receiving call response [100], dialog
is [(nil)] in state [BELLE_SIP_DIALOG_NULL]
linphone-message : channel [0x1a54400]: received [816] new bytes from
[UDP://192.168.X.GWXX:5060]:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.11.16:5060;branch=z9hG4bK.rXMlLVHoU;received=192.168.11.16;rport=5060
From: <sip:address@hidden>;tag=lZR0H0cMu
To: sip:address@hidden;tag=as315e2721
Call-ID: ugGHDKr058
CSeq: 21 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+b1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:address@hidden:5060>
Content-Type: application/sdp
Content-Length: 323

v=0
o=root 2075111992 2075111992 IN IP4 192.168.X.GWXX
s=Asterisk PBX 11.13.1~dfsg-2+b1
c=IN IP4 192.168.X.GWXX
t=0 0
m=audio 14152 RTP/AVP 96 18 3 101
a=rtpmap:96 speex/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

linphone-message : channel [0x1a54400] [493] bytes parsed
linphone-message : channel [0x1a54400] read [323] bytes of body from
[192.168.X.GWXX:5060]
linphone-message : Found transaction matching response.
linphone-message : Changing [client] [INVITE] transaction [0x1b4d3f0],
from state [PROCEEDING] to [ACCEPTED]
linphone-message : New client dialog [0x1b37ac0] , local tag
[lZR0H0cMu], remote tag [as315e2721]
linphone-message : Dialog [0x1b37ac0]: now updated by transaction
[0x1b4d3f0].
linphone-message : op [0x1afc300] : set_or_update_dialog()
current=[(nil)] new=[0x1b37ac0]
linphone-message : Op [0x1afc300] receiving call response [200], dialog
is [0x1b37ac0] in state [BELLE_SIP_DIALOG_CONFIRMED]
linphone-message : Found payload speex/8000 fmtp=
linphone-message : Found payload G729/8000 fmtp=annexb=no
linphone-message : Found payload GSM/8000 fmtp=
linphone-message : Found payload telephone-event/8000 fmtp=0-16
linphone-message : Doing SDP offer/answer process of type outgoing
linphone-message : Processing for stream 0
linphone-message : Adding G722/8000 for compatibility, just in case.
linphone-message : Adding speex/32000 for compatibility, just in case.
linphone-message : Adding telephone-event/32000 for compatibility, just
in case.
linphone-message : Computing branch id z9hG4bK.-e1PDAgLE for message
sent statelessly
linphone-message : channel [0x1a54400]: message sent to
[UDP://192.168.X.GWXX:5060], size: [415] bytes
ACK sip:address@hidden:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.16:5060;rport;branch=z9hG4bK.-e1PDAgLE
From: <sip:address@hidden>;tag=lZR0H0cMu
To: <sip:address@hidden>;tag=as315e2721
CSeq: 21 ACK
Call-ID: ugGHDKr058
Max-Forwards: 70
Authorization:  Digest realm="gw", nonce="7c80bc5f", algorithm=MD5,
username="pc7",  uri="sip:address@hidden",
response="73e29b9c7321f940641cc951a5bc0121"


linphone-message : Call 0x1afde90: moving from state
LinphoneCallOutgoingProgress to LinphoneCallConnected
linphone-message : StatusIcon: blinking set to FALSE
linphone-message : Call 0x1afde90 is locking sound resources.
linphone-message : Linphone core [0x1876220] notifying [call_state_changed]
linphone-message : Linphone core [0x1876220] notifying [display_status]
linphone-message : linphone_call_start_media_streams() call=[0x1afde90]
local upload_bandwidth=[0] kbit/s; local download_bandwidth=[0] kbit/s
linphone-message : Audio bandwidth for this call is 32
linphone-message : RtpSession [0x1b01800] sending to rtp
[192.168.X.GWXX:14152] rtcp [192.168.X.GWXX:14153]
linphone-message : Stun packet sent for session [0x1b01800]
linphone-message : ms_filter_unlink:
MSRtpRecv:0x1af0ed0,0-->MSVoidSink:0x1ae9510,0
linphone-message : speex_lib_ctl does not support SPEEX_LIB_CPU_FEATURE_NEON
linphone-message : Setting echo canceller delay with value provided by
soundcard: 0 ms
linphone-error : No such filter with id 117
linphone-message : target bitrate not set for stream [0x1663a00] using
payload's bitrate is 32000
linphone-message : Setting audio encoder network bitrate to [32000] on
stream [0x1663a00]
linphone-message : MSSpeexEnc: got ptime=20
linphone-message : MSSpeexEnc: got ptime=20
linphone-message : Equalizer rate: 8000, selecting 128 steps for FFT
linphone-message : Equalizer rate: 8000, selecting 128 steps for FFT
linphone-message : ms_filter_link:
MSPulseRead:0x1ae9510,0-->MSSpeexEC:0x1a68f40,1
linphone-message : ms_filter_link:
MSSpeexEC:0x1a68f40,1-->MSVolume:0x1b59fc0,0
linphone-message : ms_filter_link:
MSVolume:0x1b59fc0,0-->MSAudioMixer:0x1b0d9a0,0
linphone-message : ms_filter_link:
MSAudioMixer:0x1b0d9a0,0-->MSSpeexEnc:0x1b6f1d0,0
linphone-message : ms_filter_link:
MSSpeexEnc:0x1b6f1d0,0-->MSRtpSend:0x1b2b940,0
linphone-message : ms_filter_link:
MSRtpRecv:0x1b3a750,0-->MSSpeexDec:0x1b59f10,0
linphone-message : ms_filter_link:
MSSpeexDec:0x1b59f10,0-->MSDtmfGen:0x1b3a620,0
linphone-message : ms_filter_link:
MSDtmfGen:0x1b3a620,0-->MSVolume:0x1b5f800,0
linphone-message : ms_filter_link: MSVolume:0x1b5f800,0-->MSTee:0x1b6e890,0
linphone-message : ms_filter_link:
MSTee:0x1b6e890,0-->MSEqualizer:0x1b72050,0
linphone-message : ms_filter_link:
MSEqualizer:0x1b72050,0-->MSAudioMixer:0x1b4f210,0
linphone-message : speex_lib_ctl does not support SPEEX_LIB_CPU_FEATURE_NEON
linphone-message : ms_filter_link:
MSFilePlayer:0x1b6b870,0-->MSResample:0x1b6b900,0
linphone-message : ms_filter_link:
MSResample:0x1b6b900,0-->MSAudioMixer:0x1b4f210,1
linphone-message : ms_filter_link:
MSAudioMixer:0x1b4f210,0-->MSSpeexEC:0x1a68f40,0
linphone-message : ms_filter_link:
MSSpeexEC:0x1a68f40,0-->MSPulseWrite:0x1af0ed0,0
linphone-message : ms_filter_link:
MSAudioMixer:0x1b0d9a0,1-->MSAudioMixer:0x1b5cd30,0
linphone-message : ms_filter_link:
MSTee:0x1b6e890,1-->MSAudioMixer:0x1b5cd30,1
linphone-message : ms_filter_link:
MSAudioMixer:0x1b5cd30,0-->MSFileRec:0x1b56370,0
linphone-message : pulseaudio record stream connected (8000Hz, 1ch)
linphone-message : Initializing speex echo canceler with framesize=64,
filterlength=2000, delay_samples=0
linphone-message : Setting maxbitrate=16000 to speex encoder.
linphone-message : Using bitrate 15000 for speex encoder, ip bitrate is
30800
linphone-message : Initializing speex resampler in mode [voip]
linphone-message : pulseaudio playback stream connected (8000Hz, 1ch)
linphone-message : Filter MSRtpRecv is already being scheduled; nothing
to do.
linphone-error : no such method on filter MSPulseWrite, fid=16394 method
index=2
linphone-message : MSVolume set gain to [0,000000 db], [1,000000] linear
linphone-message : No valid video stream defined.
linphone-message : LinphoneCall[0x1afde90] : payload type 96 speex/8000
fmtp=vbr=on added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 18 G729/8000
fmtp=annexb=no added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 3 GSM/8000
fmtp= added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 101
telephone-event/8000 fmtp= added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 9 G722/8000
fmtp= added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 97 speex/32000
fmtp=vbr=on added to frozen list.
linphone-message : LinphoneCall[0x1afde90] : payload type 98
telephone-event/32000 fmtp= added to frozen list.
linphone-message : audio stream index found: 0, updating main audio
stream index
linphone-message : Call 0x1afde90: moving from state
LinphoneCallConnected to LinphoneCallStreamsRunning
linphone-message : Linphone core [0x1876220] notifying [call_state_changed]
linphone-message : Garbage collecting unowned object of type belle_sip_hop_t
linphone-message : Garbage collecting unowned object of type
belle_sdp_session_description_t
linphone-warning : Getting reference signal but no echo to synchronize on.
linphone-warning : Not enough ref samples, using zeroes
linphone-message : MSAudioMixer [0x1b0d9a0] is entering bypass mode.
linphone-message : Stun packet sent for session [0x1b01800]
linphone-message : Samples are back.
linphone-warning : Not enough ref samples, using zeroes
linphone-warning : Bad RTCP packet, too short.
linphone-warning : Bad RTCP packet, too short.
linphone-warning : Bad RTCP packet, too short.
linphone-warning : Bad RTCP packet, too short.

To inifinity ... every ... 50 ms ?


It seems, that linphone kill's itself funktionality due execcsive logspam.

Regards and a Happy new Year
Jürgen

> Gautier Pelloux-Prayer
> Software Engineer @ Belledonne Communications
> 
>> On 03 Jan 2016, at 22:12, Juergen Sauer <address@hidden> wrote:
>>
>> Hi,
>> I stumbled into an ugly behavior of linphone.
>> Version 3.9.1 (Arch Linux, out of official Repro)
>>
>> During a call to any number of the asteris server linphone freezes and
>> is continously freezing.
>>
>> Either any UI Action are possible, nor canceling the call is posible.
>>
>> The only way out ist killing the  process hardly.
>>
>> Any idea according this critical bug?
>>
>> (BTW, zoiper, ekiga are running fine with the same setup).
>>
>> mit freundlichen Grüßen
>> Jürgen Sauer
>> -- 
>> Jürgen Sauer - automatiX GmbH,
>> +49-4209-4699, address@hidden
>> Geschäftsführer: Jürgen Sauer,
>> Gerichtstand: Amtsgericht Walsrode • HRB 120986
>> Ust-Id: DE191468481 • St.Nr.: 36/211/08000
>> GPG Public Key zur Signaturprüfung:
>> http://www.automatix.de/juergen_sauer_publickey.gpg
>>
>> _______________________________________________
>> Linphone-users mailing list
>> address@hidden
>> https://lists.nongnu.org/mailman/listinfo/linphone-users
> 
> 
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> https://lists.nongnu.org/mailman/listinfo/linphone-users
> 


mit freundlichen Grüßen
Jürgen Sauer
-- 
Jürgen Sauer - automatiX GmbH,
+49-4209-4699, address@hidden
Geschäftsführer: Jürgen Sauer,
Gerichtstand: Amtsgericht Walsrode • HRB 120986
Ust-Id: DE191468481 • St.Nr.: 36/211/08000
GPG Public Key zur Signaturprüfung:
http://www.automatix.de/juergen_sauer_publickey.gpg