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 <[email protected]> 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 > [jojo@pc6 ~]$ 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 [100@internal:1] Answer("SIP/pc7-00000008", "") in new > stack >> 0x7fd1bc0314a0 -- Probation passed - setting RTP source address > to 192.168.11.16:7078 > -- Executing [100@internal:2] Wait("SIP/pc7-00000008", "1") in new stack > -- Executing [100@internal:3] Set("SIP/pc7-00000008", > "FUTURETIME=1451903550") in new stack > -- Executing [100@internal:4] Set("SIP/pc7-00000008", > "TIME=1451907150") in new stack > -- Executing [100@internal: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 [100@internal:6] WaitUntil("SIP/pc7-00000008", > "1451903550") in new stack > -- Executing [100@internal:7] Playback("SIP/pc7-00000008", "beep") > in new stack > -- <SIP/pc7-00000008> Playing 'beep.gsm' (language 'de') > -- Executing [100@internal: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:[email protected]>;tag=Ra~AZ9KZQ > To: sip:[email protected] > 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:[email protected]>;+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:[email protected]] 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:[email protected]>;tag=Ra~AZ9KZQ > To: sip:[email protected];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:[email protected]>;tag=Ra~AZ9KZQ > To: sip:[email protected] > 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:[email protected]>;+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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.X.GWXX:5060;branch=z9hG4bK0fc032b7 > Max-Forwards: 70 > From: "asterisk" <sip:[email protected]>;tag=as7a5bfc60 > To: <sip:[email protected]> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]: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:[email protected]>;tag=as7a5bfc60 > To: <sip:[email protected]>;tag=F1rb9 > Call-ID: [email protected]: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:[email protected]>;tag=Ra~AZ9KZQ > To: sip:[email protected];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:[email protected]>;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:[email protected]] 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:[email protected]] calling > [sip:[email protected]] 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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.jY1rWOTaq;rport > From: <sip:[email protected]>;tag=lZR0H0cMu > To: sip:[email protected] > 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:[email protected]>;+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:[email protected]>;tag=lZR0H0cMu > To: sip:[email protected];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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.jY1rWOTaq;rport > Call-ID: ugGHDKr058 > From: <sip:[email protected]>;tag=lZR0H0cMu > To: <sip:[email protected]>;tag=as0a014d94 > Contact: > <sip:[email protected]>;+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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 192.168.11.16:5060;branch=z9hG4bK.rXMlLVHoU;rport > From: <sip:[email protected]>;tag=lZR0H0cMu > To: sip:[email protected] > 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:[email protected]>;+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:[email protected]", > 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:[email protected]>;tag=lZR0H0cMu > To: sip:[email protected] > 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:[email protected]: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:[email protected]>;tag=lZR0H0cMu > To: sip:[email protected];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:[email protected]: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:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.11.16:5060;rport;branch=z9hG4bK.-e1PDAgLE > From: <sip:[email protected]>;tag=lZR0H0cMu > To: <sip:[email protected]>;tag=as315e2721 > CSeq: 21 ACK > Call-ID: ugGHDKr058 > Max-Forwards: 70 > Authorization: Digest realm="gw", nonce="7c80bc5f", algorithm=MD5, > username="pc7", uri="sip:[email protected]", > 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 <[email protected]> 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, [email protected] >>> 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 >>> [email protected] >>> https://lists.nongnu.org/mailman/listinfo/linphone-users >> >> >> _______________________________________________ >> Linphone-users mailing list >> [email protected] >> https://lists.nongnu.org/mailman/listinfo/linphone-users >> > > > mit freundlichen Grüßen > Jürgen Sauer > -- > Jürgen Sauer - automatiX GmbH, > +49-4209-4699, [email protected] > 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 > [email protected] > https://lists.nongnu.org/mailman/listinfo/linphone-users _______________________________________________ Linphone-users mailing list [email protected] https://lists.nongnu.org/mailman/listinfo/linphone-users
