Allo Emmanuel,
On Friday 08 August 2014 10:24:52 Emmanuel Lepage wrote:
> The latest packages, available at
> https://launchpad.net/~savoirfairelinux/+archive/ubuntu/sflphone-nightly/
> should fix the failed calls hang up issue.
Will check out the new version asap.
> As for the other issue, I still
> need the extra sflphoned information. As I said in my last email, the
> client log doesn't seem to indicate that the second call was ever sent. I
> also improved the log messages for this section to make tracking this
> easier.
Followed your instructions:
- Startup /usr/lib/sflphone/sflphoned manually
- Startup sflphone-client-kde
- Call non-existent number (my old landline no) using keyboard dialpad
- Hit ENTER key → busy signal
- Click _Hangup_ UI element → SFLphone doesn't hangup
- Dial 1 using keyboards number pad → busy signal is gone
- Hit ESC → SFLphone is in ready state again
Find (anonymized) log data attached.
Thanks for your help,
Florian
$ /usr/lib/sflphone/sflphoned -c -d
SFLphone Daemon 1.4.0, by Savoir-Faire Linux 2004-2014
http://www.sflphone.org/
audiocodecfactory.cpp:215:0x5980: Scanning /home/myusername/.sflphone/ to find audio codecs....
audiocodecfactory.cpp:215:0x5980: Scanning /usr/lib/sflphone/codecs/ to find audio codecs....
g729: did not open shared lib
audiocodecfactory.cpp:215:0x5980: Scanning /usr/lib/sflphone/audio/codecs/ to find audio codecs....
audiocodecfactory.cpp:62:0x5980: Loaded codec GSM
audiocodecfactory.cpp:62:0x5980: Loaded codec speex
audiocodecfactory.cpp:62:0x5980: Loaded codec speex
audiocodecfactory.cpp:62:0x5980: Loaded codec PCMA
audiocodecfactory.cpp:62:0x5980: Loaded codec speex
audiocodecfactory.cpp:62:0x5980: Loaded codec G722
audiocodecfactory.cpp:62:0x5980: Loaded codec PCMU
client.cpp:76:0x5980: DBUS init threading
client.cpp:78:0x5980: DBUS instantiate default dispatcher
client.cpp:88:0x5980: DBUS session connection to session bus
client.cpp:90:0x5980: DBUS request org.sflphone.SFLphone from session connection
client.cpp:93:0x5980: DBUS create call manager from session connection
client.cpp:95:0x5980: DBUS create configuration manager from session connection
client.cpp:97:0x5980: DBUS create presence manager from session connection
client.cpp:101:0x5980: DBUS create instance manager from session connection
video_device_monitor.cpp:155:0x5980: Manager not initialized yet
video_device_monitor.cpp:155:0x5980: Manager not initialized yet
client.cpp:110:0x5980: DBUS system connection to system bus
client.cpp:112:0x5980: DBUS create the network manager from the system bus
client.cpp:121:0x5980: DBUS registration done
manager.cpp:41:0x5980: Not initialized
managerimpl.cpp:181:0x5980: Configuration file path: /home/myusername/.config/sflphone/sflphoned.yml
sipvoiplink.cpp:629:0x5980: creating SIPVoIPLink instance
16:58:06.731 os_core_unix.c !pjlib 2.2.1 for POSIX initialized
sipvoiplink.cpp:588:0x5980: pjsip version 2.2.1 for x86_64-unknown-linux-gnu initialized
sipvoiplink.cpp:702:0x8700: Registering thread
sipaccount.cpp:1610:0x5980: Presence enabled for Account:864217638 : false.
sipaccount.cpp:1610:0x5980: Presence enabled for IP2IP : false.
sipaccount.cpp:896:0x5980: SIPAccount::registerVoIPLink
siptransport.cpp:232:0x5980: Created UDP transport on default : 0.0.0.0:5060
pulselayer.cpp:188:0x5980: Waiting....
pulselayer.cpp:188:0x7700: Waiting....
pulselayer.cpp:188:0x7700: Waiting....
pulselayer.cpp:192:0x7700: Connection to PulseAudio server established
pulselayer.cpp:749:0x7700: Sink 0
Name: alsa_output.pci-0000_00_03.0.hdmi-stereo
Driver: module-alsa-card.c
Description: Internes Audio Digital Stereo (HDMI)
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 5
Volume: muted
Monitor Source: 0
Latency: 0 usec
Flags: LATENCY HARDWARE
pulselayer.cpp:236:0x7700: seeking for alsa_output.pci-0000_00_03.0.hdmi-stereo in sinks. NOT found
pulselayer.cpp:749:0x7700: Sink 1
Name: alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
Driver: module-alsa-card.c
Description: Plantronics .Audio 648 USB Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 7
Volume: 0: 58% 1: 58%
Monitor Source: 2
Latency: 0 usec
Flags: HW_VOLUME_CTRL LATENCY HARDWARE
pulselayer.cpp:236:0x7700: seeking for alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo in sinks. NOT found
pulselayer.cpp:749:0x7700: Sink 2
Name: alsa_output.pci-0000_00_1b.0.analog-stereo
Driver: module-alsa-card.c
Description: Internes Audio Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 8
Volume: muted
Monitor Source: 4
Latency: 0 usec
Flags: HW_VOLUME_CTRL LATENCY HARDWARE
pulselayer.cpp:236:0x7700: seeking for alsa_output.pci-0000_00_1b.0.analog-stereo in sinks. NOT found
pulselayer.cpp:709:0x7700: Source 0
Name: alsa_output.pci-0000_00_03.0.hdmi-stereo.monitor
Driver: module-alsa-card.c
Description: Monitor of Internes Audio Digital Stereo (HDMI)
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 5
Volume: 0: 100% 1: 100%
Monitor if Sink: 0
Latency: 0 usec
Flags: LATENCY
pulselayer.cpp:244:0x7700: seeking for alsa_output.pci-0000_00_03.0.hdmi-stereo.monitor in sources. NOT found
pulselayer.cpp:709:0x7700: Source 1
Name: alsa_input.usb-046d_HD_Pro_Webcam_C920_06A6729F-02-C920.analog-stereo
Driver: module-alsa-card.c
Description: HD Pro Webcam C920 Analog Stereo
Sample Specification: s16le 2ch 32000Hz
Channel Map: front-left,front-right
Owner Module: 6
Volume: muted
Monitor if Sink: 4294967295
Latency: 0 usec
Flags: HW_VOLUME_CTRL LATENCY HARDWARE
pulselayer.cpp:244:0x7700: seeking for alsa_input.usb-046d_HD_Pro_Webcam_C920_06A6729F-02-C920.analog-stereo in sources. NOT found
pulselayer.cpp:709:0x7700: Source 2
Name: alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo.monitor
Driver: module-alsa-card.c
Description: Monitor of Plantronics .Audio 648 USB Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 7
Volume: 0: 100% 1: 100%
Monitor if Sink: 1
Latency: 0 usec
Flags: LATENCY
pulselayer.cpp:244:0x7700: seeking for alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo.monitor in sources. NOT found
pulselayer.cpp:709:0x7700: Source 3
Name: alsa_input.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
Driver: module-alsa-card.c
Description: Plantronics .Audio 648 USB Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 7
Volume: 0: 100% 1: 100%
Monitor if Sink: 4294967295
Latency: 0 usec
Flags: HW_VOLUME_CTRL LATENCY HARDWARE
pulselayer.cpp:244:0x7700: seeking for alsa_input.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo in sources. NOT found
pulselayer.cpp:709:0x7700: Source 4
Name: alsa_output.pci-0000_00_1b.0.analog-stereo.monitor
Driver: module-alsa-card.c
Description: Monitor of Internes Audio Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 8
Volume: 0: 100% 1: 100%
Monitor if Sink: 2
Latency: 0 usec
Flags: LATENCY
pulselayer.cpp:244:0x7700: seeking for alsa_output.pci-0000_00_1b.0.analog-stereo.monitor in sources. NOT found
pulselayer.cpp:709:0x7700: Source 5
Name: alsa_input.pci-0000_00_1b.0.analog-stereo
Driver: module-alsa-card.c
Description: Internes Audio Analog Stereo
Sample Specification: s16le 2ch 44100Hz
Channel Map: front-left,front-right
Owner Module: 8
Volume: muted
Monitor if Sink: 4294967295
Latency: 0 usec
Flags: HW_VOLUME_CTRL LATENCY HARDWARE
pulselayer.cpp:244:0x7700: seeking for alsa_input.pci-0000_00_1b.0.analog-stereo in sources. NOT found
sipaccount.cpp:896:0x5980: SIPAccount::registerVoIPLink phone.mycompany.com
sipaccount.cpp:900:0x5980: --- SIPSERVERIP
main.cpp:198:0x5980: Built with video support
managerimpl.cpp:236:0x5980: Starting client event loop
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
pulselayer.cpp:344:0x5980: playback: alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo record: alsa_input.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo ringtone: alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
audiostream.cpp:52:0x5980: SFLphone playback: trying to create stream with device alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo (48000Hz, 2 channels)
audiostream.cpp:52:0x5980: SFLphone capture: trying to create stream with device alsa_input.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo (48000Hz, 2 channels)
audiostream.cpp:52:0x5980: SFLphone ringtone: trying to create stream with device alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo (48000Hz, 2 channels)
audiolayer.cpp:61:0x5980: hardwareFormatAvailable : {2 channels, 48000kHz}
managerimpl.cpp:2207:0x5980: Audio format changed: {1 channels, 48000kHz} -> {2 channels, 48000kHz}
audiostream.cpp:122:0x7700: Stream successfully created, connected to alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
audiostream.cpp:123:0x7700: maxlength 30720
audiostream.cpp:124:0x7700: tlength 11520
audiostream.cpp:125:0x7700: prebuf 0
audiostream.cpp:126:0x7700: minreq 3840
audiostream.cpp:127:0x7700: fragsize 15360
audiostream.cpp:128:0x7700: samplespec s16le 2ch 48000Hz
audiostream.cpp:122:0x7700: Stream successfully created, connected to alsa_input.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
audiostream.cpp:123:0x7700: maxlength 30720
audiostream.cpp:124:0x7700: tlength 15360
audiostream.cpp:125:0x7700: prebuf 0
audiostream.cpp:126:0x7700: minreq 4294967295
audiostream.cpp:127:0x7700: fragsize 7680
audiostream.cpp:128:0x7700: samplespec s16le 2ch 48000Hz
audiostream.cpp:122:0x7700: Stream successfully created, connected to alsa_output.usb-Plantronics_Plantronics_.Audio_648_USB-00-USB.analog-stereo
audiostream.cpp:123:0x7700: maxlength 30720
audiostream.cpp:124:0x7700: tlength 11520
audiostream.cpp:125:0x7700: prebuf 0
audiostream.cpp:126:0x7700: minreq 3840
audiostream.cpp:127:0x7700: fragsize 15360
audiostream.cpp:128:0x7700: samplespec s16le 2ch 48000Hz
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
sipvoiplink.cpp:1276:0x5980: No SIP call with ID 965027506
managerimpl.cpp:331:0x5980: New outgoing call 965027506 to 0761123456789
managerimpl.cpp:352:0x5980: Selecting account Account:864217638
sipvoiplink.cpp:882:0x5980: New outgoing call to 0761123456789
sipvoiplink.cpp:958:0x5980: UserAgent: New registered account call to 0761123456789
audiorecord.cpp:97:0x5980: Generate filename for this call 20140814-16:59:25
recordable.cpp:36:0x5980: Set recording options: /home/myusername
audio_symmetric_rtp_session.cpp:48:0x5980: Setting new RTP session with destination 192.168.20.75:21370
audio_rtp_session.cpp:281:0x5980: Preparing receiving thread
audio_rtp_session.cpp:244:0x5980: Set session scheduling timeout (4000) and expireTimeout (1000000)
audio_rtp_session.cpp:107:0x5980: Switching to a transport rate of 20 ms
audiorecord.cpp:145:0x5980: Concatenate .wav file extension: name : 20140814-16:59:25
sdp.cpp:438:0x5980: No selected video codec while building local SDP offer
sdp.cpp:518:0x5980: SDP: Local SDP Session:
sdp.cpp:468:0x5980: v=0
o=localhost 3617017165 0 IN IP4 192.168.20.75
s=sflphone
c=IN IP4 192.168.20.75
t=0 0
m=audio 21370 RTP/AVP 9 0 8 3 110 111 112 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:21371 IN IP4 192.168.20.75
sipvoiplink.cpp:1574:0x5980: contact header: <sip:[email protected]:5060> / <sip:[email protected]> -> <sip:[email protected]>
managerimpl.cpp:306:0x5980: ----- Switch current call id to 965027506 -----
managerimpl.cpp:306:0x8700: ----- Switch current call id to -----
sipvoiplink.cpp:1256:0x8700: Removing call 965027506 from list
sipvoiplink.cpp:1276:0x5980: No SIP call with ID 965027506
managerimpl.cpp:2664:0x5980: Call is NULL
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
sipvoiplink.cpp:646:0x8700: username = MySIPuser, server = SIPSERVERIP
sipaccount.cpp:1650:0x8700: Matching account id in request is a fullmatch MySIPuser@SIPSERVERIP
managerimpl.cpp:468:0x5980: Send call state change (HUNGUP) for id 1115753520
sipvoiplink.cpp:1276:0x5980: No SIP call with ID 1115753520
managerimpl.cpp:473:0x5980: Could not hang up call 1115753520, call not valid
_______________________________________________
SFLphone mailing list
[email protected]
http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
