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

[Gautier Pelloux-Prayer]

Can you run Linphone under gdb, and hit Ctrl+C when Linphone is frozen please?

Please send the full output of "bt full".

Thanks,

Gautier Pelloux-Prayer
Software Engineer @ Belledonne Communications

> On 04 Jan 2016, at 12:37, Juergen Sauer <address@hidden> wrote:
> 
> 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
> 
> 
> _______________________________________________
> Linphone-users mailing list
> address@hidden
> https://lists.nongnu.org/mailman/listinfo/linphone-users