URGENT - pjsip 2.1 - audio playback, audio capture do not work at all its being 4 weeks now any advise on this please?

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Hi ,

Seems like u r nt usng any proxy server 

chk
./pjsua-x86_64-unknown-linux-gnu --help is thr any option for no
registeration and still establishing call. 

plz try switch
--reg-use-proxy=0 then chk whtr u r able to establish call.

Thanks,
Gaurav

On Tue, 3 Dec 2013 13:34:32 +0100, Shamun Toha Md 
wrote:
> Hello
*Gaurav, Varun, Dennis, Nishant*
> 
> Can you please tell me why after
installing pjsip 2.1 perfectly, with
> libasound2 and all , i still do not
have audio playback? (i checked with
> speaker-test, alsa sink, src
mplayer, vlc, ffmpeg my speaker and mic is
> available without pjsip it
works, but with pjsip i still do not hear any
> single audio playback, also
when i am connected i have no microphone
> capture)
> 
> Please can you
kindly share, i have been trying this for about now 4 weeks,
> still its
not working at all.
> 
> Please see the details of following steps how i
installed it and how i
> tested it.
> 
> *Step 1*: install and run
> 
> $
cd /var/tmp
> $ wget
http://www.pjsip.org/release/2.1/pjproject-2.1.tar.bz2
> $ tar xvfj
pjproject-2.1.tar.bz2
> $ cd pjproject-2.1
> $ ./configure
> $ make dep &&
make && make install
> 
> 
> # Python enable (optional)
> $ cd
/var/tmp/pjproject-2.1.0/pjsip-apps/src/python
> $ python setup.py
install
> $ python
> Python 2.7.5+ (default, Sep 19 2013, 13:48:49)
> [GCC
4.8.1] on linux2
> Type "help", "copyright", "credits" or "license" for
more information.
>>>> import pjsua
>>>>
> 
> 
> *Step 2*: Basic kick start
sample to register and make call, by manually
> assigning playback id and
capture id , this also do not work for audio
> capture and playback:
https://gist.github.com/anonymous/7768285
> 
> Here you can see i used the
latest release built in pjsua which also giving
> no sound and no luck to
capture microphone.
> 
> $ ./pjsua-x86_64-unknown-linux-gnu
> 13:24:33.632
os_core_unix.c !pjlib 2.1 for POSIX initialized
> 13:24:33.632
sip_endpoint.c .Creating endpoint instance...
> 13:24:33.633 pjlib
.select() I/O Queue created (0x20fb8a0)
> 13:24:33.633 sip_endpoint.c
.Module "mod-msg-print" registered
> 13:24:33.633 sip_transport. .Transport
manager created.
> 13:24:33.633 pjsua_core.c .PJSUA state changed: NULL -->
CREATED
> 13:24:33.633 sip_endpoint.c .Module "mod-pjsua-log" registered
>
13:24:33.633 sip_endpoint.c .Module "mod-tsx-layer" registered
>
13:24:33.633 sip_endpoint.c .Module "mod-stateful-util" registered
>
13:24:33.633 sip_endpoint.c .Module "mod-ua" registered
> 13:24:33.633
sip_endpoint.c .Module "mod-100rel" registered
> 13:24:33.633
sip_endpoint.c .Module "mod-pjsua" registered
> 13:24:33.633 sip_endpoint.c
.Module "mod-invite" registered
> bt_audio_service_open: connect() failed:
Connection refused (111)
> bt_audio_service_open: connect() failed:
Connection refused (111)
> bt_audio_service_open: connect() failed:
Connection refused (111)
> bt_audio_service_open: connect() failed:
Connection refused (111)
> 13:24:33.702 pa_dev.c ..PortAudio sound library
initialized,
> status=0
> 13:24:33.702 pa_dev.c ..PortAudio host api
count=2
> 13:24:33.702 pa_dev.c ..Sound device count=20
> 13:24:33.702
pjlib ..select() I/O Queue created (0x21579f8)
> 13:24:33.711
sip_endpoint.c .Module "mod-evsub" registered
> 13:24:33.711 sip_endpoint.c
.Module "mod-presence" registered
> 13:24:33.711 sip_endpoint.c .Module
"mod-mwi" registered
> 13:24:33.711 sip_endpoint.c .Module "mod-refer"
registered
> 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-pres"
registered
> 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-im"
registered
> 13:24:33.711 sip_endpoint.c .Module "mod-pjsua-options"
registered
> 13:24:33.711 pjsua_core.c .1 SIP worker threads created
>
13:24:33.711 pjsua_core.c .pjsua version 2.1 for Linux-3.11.0.12/x86_64/
>
glibc-2.17 initialized
> 13:24:33.711 pjsua_core.c .PJSUA state changed:
CREATED --> INIT
> 13:24:33.711 sip_endpoint.c Module "mod-default-handler"
registered
> 13:24:33.711 pjsua_core.c bind() error: Address already in use
[status=
> 120098]
> 13:24:33.711 pjsua_core.c Shutting down, flags=0...
>
13:24:33.711 pjsua_core.c PJSUA state changed: INIT --> CLOSING
>
13:24:33.721 pjsua_call.c .Hangup all calls..
> 13:24:33.721 pjsua_pres.c
.Shutting down presence..
> 13:24:33.721 pjsua_media.c .Shutting down
media..
> 13:24:33.721 pjsua_media.c ..Call 0: deinitializing media..
>
13:24:33.721 pjsua_media.c ..Call 1: deinitializing media..
> 13:24:33.721
pjsua_media.c ..Call 2: deinitializing media..
> 13:24:33.721 pjsua_media.c
..Call 3: deinitializing media..
> 13:24:34.203 pa_dev.c ..PortAudio sound
library shutting down..
> 13:24:35.210 pjsua_core.c .Destroying...
>
13:24:35.210 sip_transactio .Stopping transaction layer module
>
13:24:35.210 sip_transactio .Stopped transaction layer module
>
13:24:35.210 sip_endpoint.c .Module "mod-default-handler" unregistered
>
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-options" unregistered
>
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-im" unregistered
>
13:24:35.210 sip_endpoint.c .Module "mod-pjsua-pres" unregistered
>
13:24:35.210 sip_endpoint.c .Module "mod-pjsua" unregistered
> 13:24:35.210
sip_endpoint.c .Module "mod-stateful-util" unregistered
> 13:24:35.210
sip_endpoint.c .Module "mod-refer" unregistered
> 13:24:35.210
sip_endpoint.c .Module "mod-mwi" unregistered
> 13:24:35.210 sip_endpoint.c
.Module "mod-presence" unregistered
> 13:24:35.210 sip_endpoint.c .Module
"mod-evsub" unregistered
> 13:24:35.210 sip_endpoint.c .Module "mod-invite"
unregistered
> 13:24:35.210 sip_endpoint.c .Module "mod-100rel"
unregistered
> 13:24:35.210 sip_endpoint.c .Module "mod-ua" unregistered
>
13:24:35.210 sip_transactio .Transaction layer module destroyed
>
13:24:35.210 sip_endpoint.c .Module "mod-tsx-layer" unregistered
>
13:24:35.210 sip_endpoint.c .Module "mod-msg-print" unregistered
>
13:24:35.211 sip_endpoint.c .Module "mod-pjsua-log" unregistered
>
13:24:35.211 sip_endpoint.c .Endpoint 0x20f0b08 destroyed
> 13:24:35.211
pjsua_core.c .PJSUA state changed: CLOSING --> NULL
> 13:24:35.211
pjsua_core.c .PJSUA destroyed...
>
sun at sun-Alienware-X51:/var/tmp/pjproject-2.1.0/pjsip-apps/bin$ ./pjsua-
>
x86_64-unknown-linux-gnu
> 13:24:51.994 os_core_unix.c !pjlib 2.1 for POSIX
initialized
> 13:24:51.995 sip_endpoint.c .Creating endpoint instance...
>
13:24:51.995 pjlib .select() I/O Queue created (0x9d98a0)
> 13:24:51.995
sip_endpoint.c .Module "mod-msg-print" registered
> 13:24:51.995
sip_transport. .Transport manager created.
> 13:24:51.995 pjsua_core.c
.PJSUA state changed: NULL --> CREATED
> 13:24:51.995 sip_endpoint.c
.Module "mod-pjsua-log" registered
> 13:24:51.995 sip_endpoint.c .Module
"mod-tsx-layer" registered
> 13:24:51.995 sip_endpoint.c .Module
"mod-stateful-util" registered
> 13:24:51.995 sip_endpoint.c .Module
"mod-ua" registered
> 13:24:51.995 sip_endpoint.c .Module "mod-100rel"
registered
> 13:24:51.995 sip_endpoint.c .Module "mod-pjsua" registered
>
13:24:51.995 sip_endpoint.c .Module "mod-invite" registered
>
bt_audio_service_open: connect() failed: Connection refused (111)
>
bt_audio_service_open: connect() failed: Connection refused (111)
>
bt_audio_service_open: connect() failed: Connection refused (111)
>
bt_audio_service_open: connect() failed: Connection refused (111)
>
13:24:52.013 pa_dev.c ..PortAudio sound library initialized,
> status=0
>
13:24:52.013 pa_dev.c ..PortAudio host api count=2
> 13:24:52.013 pa_dev.c
..Sound device count=20
> 13:24:52.013 pjlib ..select() I/O Queue created
(0xa359f8)
> 13:24:52.016 sip_endpoint.c .Module "mod-evsub" registered
>
13:24:52.016 sip_endpoint.c .Module "mod-presence" registered
>
13:24:52.017 sip_endpoint.c .Module "mod-mwi" registered
> 13:24:52.017
sip_endpoint.c .Module "mod-refer" registered
> 13:24:52.017 sip_endpoint.c
.Module "mod-pjsua-pres" registered
> 13:24:52.017 sip_endpoint.c .Module
"mod-pjsua-im" registered
> 13:24:52.017 sip_endpoint.c .Module
"mod-pjsua-options" registered
> 13:24:52.017 pjsua_core.c .1 SIP worker
threads created
> 13:24:52.017 pjsua_core.c .pjsua version 2.1 for
Linux-3.11.0.12/x86_64/
> glibc-2.17 initialized
> 13:24:52.017
pjsua_core.c .PJSUA state changed: CREATED --> INIT
> 13:24:52.017
sip_endpoint.c Module "mod-default-handler" registered
> 13:24:52.017
pjsua_core.c SIP UDP socket reachable at 192.168.1.19:5060
> 13:24:52.017
udp0xa4e6e0 SIP UDP transport started, published address
> is
>
192.168.1.19:5060
> 13:24:52.017 pjsua_acc.c Adding account: id=
>
13:24:52.017 pjsua_acc.c .Account added with id
> 0
> 13:24:52.017
pjsua_acc.c Acc 0: setting online status to 1..
> 13:24:52.017 tcplis:5060
SIP TCP listener ready for incoming
> connections at 192.168.1.19:5060
>
13:24:52.017 pjsua_acc.c Adding account: id= transport=TCP>
> 13:24:52.017
pjsua_acc.c .Account 
> added with id 1
> 13:24:52.017 pjsua_acc.c Acc 1:
setting online status to 1..
> 13:24:52.017 pjsua_core.c PJSUA state
changed: INIT --> STARTING
> 13:24:52.017 sip_endpoint.c .Module
"mod-unsolicited-mwi" registered
> 13:24:52.017 pjsua_core.c .PJSUA state
changed: STARTING --> RUNNING
>>>>>
> Account list:
> [ 0] : does not
register
> Online status: Online
> *[ 1] : does not register
> Online
status: Online
> Buddy list:
> -none-
> 
> 
>
+=============================================================================+
>
| Call Commands: | Buddy, IM & Presence: | Account:
> |
> | | |
> |
> | m
Make new call | +b Add new buddy .| +a Add new
> accnt |
> | M Make
multiple calls | -b Delete buddy | -a Delete
> accnt
> . |
> | a Answer
call | i Send IM | !a Modify
> accnt
> . |
> | h Hangup call (ha=all) | s
Subscribe presence | rr (Re-)
> register |
> | H Hold call | u Unsubscribe
presence | ru Unregister
> |
> | v re-inVite (release hold) | t ToGgle
Online status | > Cycle next
> ac.|
> | U send UPDATE | T Set online status
| < Cycle prev
> ac.|
> | ],[ Select next/prev call
>
+--------------------------+-------------------+
> | x Xfer call | Media
Commands: | Status &
> Config
> : |
> | X Xfer with Replaces | |
> |
> | #
Send RFC 2833 DTMF | cl List ports | d Dump status
> |
> | * Send DTMF with
INFO | cc Connect port | dd Dump
> detailed |
> | dq Dump curr. call
quality | cd Disconnect port | dc Dump config
> |
> | | V Adjust audio
Volume | f Save config
> |
> | S Send arbitrary REQUEST | Cp Codec
priorities |
> |
>
+-----------------------------------------------------------------------------+
>
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type
> |
>
+=============================================================================+
>
You have 0 active call
> 
> 
> 
> 
>>>> m
> (You currently have 0 calls)
>
Buddy list:
> -none-
> 
> 
> Choices:
> 0 For current dialog.
> -1 All 0
buddies in buddy list
> [1 - 0] Select from buddy list
> URL An URL
> Empty
input (or 'q') to cancel
> Make call: sip:9198 at 192.168.1.12
> 13:25:48.064
pjsua_call.c Making call with acc #1 to
> sip:9198 at 192.168.1.12
>
13:25:48.065 pjsua_aud.c .Set sound device: capture=-1, playback=-2
>
13:25:48.065 pjsua_app.c ..Turning sound device ON
> 13:25:48.065
pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms
> Expression
'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
>
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
>
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
>
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
>
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
>
Expression 'PaAlsaStream_Configure( stream, inputParameters,
>
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
>
&outputLatency, &hostBufferSizeMode )' failed in
>
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1994
>
Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
>
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1294
>
Expression 'PaAlsaStreamComponent_InitialConfigure( &self->capture,
>
inParams, self->primeBuffers, hwParamsCapture, &realSr )' failed in
>
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1870
>
Expression 'PaAlsaStream_Configure( stream, inputParameters,
>
outputParameters, sampleRate, framesPerBuffer, &inputLatency,
>
&outputLatency, rport;branch=
> z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
>
Max-Forwards: 70
> From: ;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
> To:
sip:9198 at 192.168.1.12
> Contact: 
> Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
> CSeq: 18614 INVITE
> Allow: PRACK,
INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER,
MESSAGE, OPTIONS
> Supported: replaces, 100rel, timer, norefersub
>
Session-Expires: 1800
> Min-SE: 90
> User-Agent: PJSUA v2.1
Linux-3.11.0.12/x86_64/glibc-2.17
> Content-Type: application/sdp
>
Content-Length: 475
> 
> 
> v=0
> o=- 3595062348 3595062348 IN IP4
192.168.1.19
> s=pjmedia
> b=AS:84
> t=0 0
> a=X-nat:0
> m=audio 40000
RTP/AVP 98 97 99 104 3 0 8 9 96
> c=IN IP4 192.168.1.19
> b=TIAS:64000
>
a=rtcp:40001 IN IP4 192.168.1.19
> a=sendrecv
> a=rtpmap:98 speex/16000
>
a=rtpmap:97 speex/8000
> a=rtpmap:99 speex/32000
> a=rtpmap:104 iLBC/8000
>
a=fmtp:104 mode=30
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
>
a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:96
telephone-event/8000
> a=fmtp:96 0-15
> 
> 
> --end msg--
> 13:25:48.123
pjsua_app.c .......Call 0 state changed to CALLING
>>>> 13:25:48.124
pjsua_core.c .RX 365 bytes Response msg
>>>> 100/INVITE/cseq=
> 18614
(rdata0xa4fd48) from UDP 192.168.1.12:5060:
> SIP/2.0 100 Trying
> Via:
SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
>
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
> From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
> To: 
> Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
> CSeq: 18614 INVITE
> User-Agent:
FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
>
Content-Length: 0
> 
> 
> 
> 
> --end msg--
> 13:25:48.144 os_core_unix.c
Info: possibly re-registering existing thread
> 13:25:48.145 pjsua_core.c
.RX 882 bytes Response msg 407/INVITE/cseq=
> 18614 (rdata0x7f5b40002998)
from UDP 192.168.1.12:5060:
> SIP/2.0 407 Proxy Authentication Required
>
Via: SIP/2.0/UDP 192.168.1.19:5060;rport=5060;branch=
>
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
> From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
> To: ;tag=DZ4am8m4t08Xr
> Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
> CSeq: 18614 INVITE
> User-Agent:
FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
> Accept:
application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer,
precondition, path, replaces
> Allow-Events: talk, hold, conference,
presence, as-feature-event, dialog,
> line-seize, call-info, sla,
include-session-description, presence.winfo,
> message-summary, refer
>
Proxy-Authenticate: Digest realm="192.168.1.19", nonce=
>
"db2a6c3c-5c29-11e3-a388-3586b66a1730", algorithm=MD5, qop="auth"
>
Content-Length: 0
> 
> 
> 
> 
> --end msg--
> 13:25:48.145 pjsua_core.c
..TX 334 bytes Request msg ACK/cseq=18614 (
> tdta0x7f5b400008c0) to UDP
192.168.1.12:5060:
> ACK sip:9198 at 192.168.1.12 SIP/2.0
> Via: SIP/2.0/UDP
192.168.1.19:5060;rport;branch=
>
z9hG4bKPjRpOevqpjWxD1o56Mw8kLUSAzT2RGX6ig
> Max-Forwards: 70
> From:
;tag=gQIWxs6E2ySbCbJ-hFrofB0SCHJwCMma
> To:
sip:9198 at 192.168.1.12;tag=DZ4am8m4t08Xr
> Call-ID:
aFBwgwgOw6gqiLvBJ4SFY73qJsqCxc7D
> CSeq: 18614 ACK
> Content-Length: 0
> 
>

> 
> 
> --end msg--
> 13:25:48.145 sip_auth_clien ....Unable to set auth
for tdta0xadcbd0: can
> not find credential for 192.168.1.19/Digest
>
13:25:48.145 pjsua_app.c .....Call 0 is DISCONNECTED [reason=407 (Proxy
>
Authentication Required)]
> 13:25:48.145 pjsua_app.c .....
> [DISCONNCTD]
To: sip:9198 at 192.168.1.12
> Call time: 00h:00m:00s, 1st res in 23 ms, conn
in 0ms
> 13:25:48.145 pjsua_media.c .....Call 0: deinitializing media..
>
13:25:49.146 pjsua_aud.c !Closing sound device after idle for 1
>
second(s)
> 13:25:49.146 pjsua_app.c .Turning sound device OFF
>
13:25:49.146 pjsua_aud.c .Closing HDA Intel PCH: ALC892 Analog
> (hw:0,0)
sound
> playback device and HDA Intel PCH: ALC892 Analog (hw:0,0) sound
capture
> device
> 
> 
> 
> 
> q
> 13:26:32.391 pjsua_core.c !Shutting
down, flags=0...
> 13:26:32.391 pjsua_core.c PJSUA state changed: RUNNING
--> CLOSING
> 13:26:32.396 pjsua_call.c .Hangup all calls..
> 13:26:32.396
pjsua_pres.c .Shutting down presence..
> 13:26:32.396 pjsua_media.c
.Shutting down media..
> 13:26:32.396 pjsua_media.c ..Call 0:
deinitializing media..
> 13:26:32.396 pjsua_media.c ..Call 1:
deinitializing media..
> 13:26:32.396 pjsua_media.c ..Call 2:
deinitializing media..
> 13:26:32.396 pjsua_media.c ..Call 3:
deinitializing media..
> 13:26:32.524 pa_dev.c ..PortAudio sound library
shutting down..
> 13:26:33.532 pjsua_core.c .Destroying...
> 13:26:33.532
sip_transactio .Stopping transaction layer module
> 13:26:33.532
sip_transactio .Stopped transaction layer module
> 13:26:33.532
sip_endpoint.c .Module "mod-default-handler" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-unsolicited-mwi" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-pjsua-options" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-pjsua-im" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-pjsua-pres" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-pjsua" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-stateful-util" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-refer" unregistered
> 13:26:33.532
sip_endpoint.c .Module "mod-mwi" unregistered
> 13:26:33.532 sip_endpoint.c
.Module "mod-presence" unregistered
> 13:26:33.532 sip_endpoint.c .Module
"mod-evsub" unregistered
> 13:26:33.532 sip_endpoint.c .Module "mod-invite"
unregistered
> 13:26:33.532 sip_endpoint.c .Module "mod-100rel"
unregistered
> 13:26:33.532 sip_endpoint.c .Module "mod-ua" unregistered
>
13:26:33.532 sip_transactio .Transaction layer module destroyed
>
13:26:33.532 sip_endpoint.c .Module "mod-tsx-layer" unregistered
>
13:26:33.532 sip_endpoint.c .Module "mod-msg-print" unregistered
>
13:26:33.532 sip_endpoint.c .Module "mod-pjsua-log" unregistered
>
13:26:33.533 tcplis:5060 .SIP TCP listener destroyed
> 13:26:33.533
sip_endpoint.c .Endpoint 0x9ceb08 destroyed
> 13:26:33.533 pjsua_core.c
.PJSUA state changed: CLOSING --> NULL
> 13:26:33.533 pjsua_core.c .PJSUA
destroyed...
> 
> 
> 
> 
> 
> 
> 
> Thank you
> Regards
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