Bug#675587: linphone: No sound at all although ALSA ok
yelloprotoss
yellowprotoss at gmail.com
Sat Jun 2 10:40:14 UTC 2012
Package: linphone
Version: 3.3.2-1
Severity: normal
Hi,
I have followed this howto and adapted it to make
http://www.kryogenix.org/days/2005/11/19/using-linphone-on-ubuntu-breezy-with-sipgate
sip:500 at ekiga.net working but no sound at all, althogh test oldphone.wav in settings works
but but
no sound.
it looks like it can see my pc since echo 500 ekiga hangs on
I also installed apt-get install speex
Please make sure that linphone-3 works. Would be so great
[sound]
playback_dev_id=ALSA: HDA Intel
ringer_dev_id=ALSA: HDA Intel
capture_dev_id=ALSA: AK5370
remote_ring=/usr/share/sounds/linphone/rings/oldphone.wav
local_ring=/usr/loca/share/sounds/linphone/rings/oldphone.wav
echocancellation=1
ec_delay=0
ec_tail_len=60
ec_frame_size=128
mic_gain=1.0
playback_gain_db=0.0
[video]
size=cif
enabled=0
display=0
capture=0
show_local=0
self_view=1
device=V4L2: /dev/video0
[net]
download_bw=0
upload_bw=0
firewall_policy=1
mtu=0
#STUN=2
stun_server=stun.ekiga.net
#Firewall=1
#nat_address=80.112.33.11
nat_address=XXXXXX
[sip]
sip_port=5060
guess_hostname=1
contact="MYUSER" <sip:MYUSER at XXXXXX>
inc_timeout=15
use_info=0
use_rfc2833=0
use_ipv6=0
register_only_when_network_is_up=1
default_proxy=-1
keepalive_period=10
only_one_codec=0
ping_with_options=1
auto_net_state_mon=1
[rtp]
audio_rtp_port=5060
video_rtp_port=5060
audio_jitt_comp=60
video_jitt_comp=0
nortp_timeout=30
[audio_codec_0]
mime=speex
rate=8000
enabled=1
recv_fmtp=vad=on
[video_codec_0]
mime=MP4V-ES
rate=90000
enabled=1
recv_fmtp=profile-level-id=3
[friend_0]
url=Alice
pol=accept
subscribe=1
[call_log_0]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:501 at ekiga.net>
start_date=Sat Jun 2 12:34:37 2012
duration=11
[call_log_1]
dir=0
status=1
from="MYUSER" <sip:MYUSER at XXXXIPXXXX>
to=<sip:613 at fwd.pulver.com>
start_date=Sat Jun 2 12:34:10 2012
duration=17
[audio_codec_1]
mime=speex
rate=16000
enabled=1
[audio_codec_2]
mime=speex
rate=32000
enabled=1
[audio_codec_3]
mime=GSM
rate=8000
enabled=1
[audio_codec_4]
mime=PCMU
rate=8000
enabled=1
[audio_codec_5]
mime=PCMA
rate=8000
enabled=1
[video_codec_1]
mime=theora
rate=90000
enabled=1
[video_codec_2]
mime=H263-1998
rate=90000
enabled=1
recv_fmtp=CIF=1;QCIF=1
[video_codec_3]
mime=H263
rate=90000
enabled=1
[video_codec_4]
mime=x-snow
rate=90000
enabled=1
[call_log_2]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:33:42 2012
duration=17
[call_log_3]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:32:57 2012
duration=39
[call_log_4]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:32:52 2012
duration=3
[call_log_5]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:32:25 2012
duration=25
[call_log_6]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:31:43 2012
duration=41
[call_log_7]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:30:55 2012
duration=3
[call_log_8]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:30:04 2012
duration=49
[GtkUi]
advanced_ui=1
uri0=sip:501 at ekiga.net
uri1=sip:613 at fwd.pulver.com
uri2=sip:500 at ekiga.net
[call_log_9]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:29:41 2012
duration=11
[call_log_10]
dir=0
status=0
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:29:15 2012
duration=21
[call_log_11]
dir=0
status=1
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:28:26 2012
duration=9
[call_log_12]
dir=0
status=1
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:26:31 2012
duration=10
[call_log_13]
dir=0
status=1
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:25:36 2012
duration=11
[call_log_14]
dir=0
status=1
from=<sip:MYUSER at ekiga.net>
to=<sip:500 at ekiga.net>
start_date=Sat Jun 2 12:25:22 2012
duration=8
[proxy_0]
reg_proxy=sip:ekiga.net
reg_route=sip:ekiga.net
reg_identity=sip:MYUSER at ekiga.net
reg_expires=120
reg_sendregister=1
publish=1
dial_escape_plus=1
[auth_info_0]
username=MYUSER
userid=MYUSER
passwd=XXXXXXXXXXXXXXXXXX
realm="ekiga.net"
Kidn regards
-- System Information:
Debian Release: 6.0.2
APT prefers stable-updates
APT policy: (500, 'stable-updates'), (500, 'stable')
Architecture: i386 (i686)
Kernel: Linux 2.6.32-5-686 (SMP w/2 CPU cores)
Locale: LANG=C, LC_CTYPE=C (charmap=ANSI_X3.4-1968)
Shell: /bin/sh linked to /bin/dash
Versions of packages linphone depends on:
ii libasound2 1.0.23-2.1 shared library for ALSA applicatio
ii libatk1.0-0 1.30.0-1 The ATK accessibility toolkit
ii libavcodec52 4:0.5.2-6 ffmpeg codec library
ii libc6 2.11.3-3 Embedded GNU C Library: Shared lib
ii libcairo2 1.8.10-6 The Cairo 2D vector graphics libra
ii libfontconfig1 2.8.0-2.1 generic font configuration library
ii libfreetype6 2.4.2-2.1+squeeze4 FreeType 2 font engine, shared lib
ii libglade2-0 1:2.6.4-1 library to load .glade files at ru
ii libglib2.0-0 2.24.2-1 The GLib library of C routines
ii libgsm1 1.0.13-3 Shared libraries for GSM speech co
ii libgtk2.0-0 2.20.1-2 The GTK+ graphical user interface
ii liblinphone3 3.3.2-1 linphone's shared library part (su
ii libmediastreamer0 3.3.2-1 linphone web phone's media library
ii libogg0 1.2.0~dfsg-1 Ogg bitstream library
ii libortp8 3.3.2-1 Real-time Transport Protocol stack
ii libpango1.0-0 1.28.3-1+squeeze2 Layout and rendering of internatio
ii libsdl1.2debian 1.2.14-6.1 Simple DirectMedia Layer
ii libspeex1 1.2~rc1-1 The Speex codec runtime library
ii libspeexdsp1 1.2~rc1-1 The Speex extended runtime library
ii libswscale0 4:0.5.2-6 ffmpeg video scaling library
ii libtheora0 1.1.1+dfsg.1-3 The Theora Video Compression Codec
ii libx11-6 2:1.3.3-4 X11 client-side library
ii libxml2 2.7.8.dfsg-2+squeeze1 GNOME XML library
ii linphone-nox 3.3.2-1 SIP softphone - console-only clien
linphone recommends no packages.
Versions of packages linphone suggests:
ii yelp 2.30.1+webkit-1 Help browser for GNOME
-- no debconf information
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