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Re: [Linphone-users]Speech path could not be through...
|
From: |
Simon Morlat |
|
Subject: |
Re: [Linphone-users]Speech path could not be through... |
|
Date: |
22 Jan 2003 10:25:42 +0100 |
Hi,
The via82cxxx_audio is known to be buggy when used by linphone.
Replace this driver by its equivalent alsa-driver
(http://www.alsa-project.org ). If your linux distribution provides
binary packages for those driver, it is easier.
Simon
Le mer 22/01/2003 à 06:23, Kanika Garg a écrit :
> Hi,
>
> I also think that the problem might be of the audio
> driver. As I am new to Linux, I am not able to trace
> the fault.
> I am hereby attaching the trace of the linphone on
> both the pc's as:
> Linphone-pc#.trace
>
> Also attached the result of /sbin/lsmod as:
> lsmod-pc.output
>
> Regards,
> Kanika Garg
>
> --- Simon Morlat <address@hidden> wrote:
> > Hi,
> >
> > I'm pretty sure that the problem you have is caused
> > of a bug in the
> > audio driver.
> > First, send me the result of /sbin/lsmod for both
> > machines so that I
> > check that you are not using a buggy driver.
> > If this is the case, then you will have to replace
> > them by alsa-drivers
> > http://www.alsa-project.org
> > Simon
> >
> > Le mar 21/01/2003 à 06:52, Kanika Garg a écrit :
> > > Hi all,
> > >
> > > I am facing some problem while putting a call
> > through
> > > using two instances of Linphone running on 2
> > different
> > > PCs on LAN.
> > >
> > > The parameters that have been set are -
> > >
> > > PC 1
> > > ----
> > >
> > > RTP port - 7000
> > > SIP port - 5060
> > > URL - sip:address@hidden
> > >
> > >
> > > PC 2
> > > ----
> > >
> > > RTP port - 7000
> > > SIP port - 5060
> > > URL - sip:address@hidden
> > >
> > > The interface used was eth0. I selected speex
> > codec
> > > and audio input was mic. And I did not select the
> > > checkbox to kill sound deamons, as specified in
> > the
> > > Linphone's user manual. When I initiate a call
> > from PC
> > > 1, it shows in the status bar,
> > >
> > > Contacting <sip:address@hidden:5060>
> > >
> > > and on the other PC, it shows that -
> > > sip:address@hidden:5060 is calling you.....
> > >
> > > and I am able to hear the ring tone. But when the
> > > incoming call is answered on PC 2, speech path is
> > not
> > > put through. I can only hear some sound, some
> > noise
> > > and not what is being said on the mic, on the
> > other
> > > hand. Can anyone help me out in this regard... If
> > > required, I can attach the output/trace.
> > >
> > > Thanking you,
> > >
> > > Regards,
> > > Kanika Garg
> > > HFCL, R&D Division,
> > > Gurgaon
> > >
> > >
> > > __________________________________________________
> > > Do you Yahoo!?
> > > Yahoo! Mail Plus - Powerful. Affordable. Sign up
> > now.
> > > http://mailplus.yahoo.com
> > >
> > >
> > > _______________________________________________
> > > Linphone-users mailing list
> > > address@hidden
> > >
> >
> http://mail.nongnu.org/mailman/listinfo/linphone-users
> > --
> > Simon Morlat <address@hidden>
> >
>
> __________________________________________________
> Do you Yahoo!?
> Yahoo! Mail Plus - Powerful. Affordable. Sign up now.
> http://mailplus.yahoo.com
>
> __________________________________________________
> Do you Yahoo!?
> Yahoo! Mail Plus - Powerful. Affordable. Sign up now.
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>
> ______________________________________________________________________
>
> 1) Starting linphone
>
>
> address@hidden /root]# linphone
> | INFO1 | <osipua.c: 59> Starting osip stack and osipua layer.
>
> | INFO1 | <udp.c: 76> Entering osipua thread.
>
> MediaStreamer-Message: Detected /dev/dsp -
>
> MediaStreamer-WARNING **: Unable to find mixer for id 1.
> MediaStreamer-Message: Detected /dev/dsp1 -
>
> MediaStreamer-WARNING **: Unable to find mixer for id 3.
> Found 2 interfaces.
> Found eth0 interface with ip address 192.168.8.70
>
> GnomeUI-WARNING **: Could not open help topics file NULL
> Adding <sip:address@hidden> to the list of alias.
> Message: Added audio payload type to SDP properties 110 speex-4/8000/1
> Message: Added audio payload type to SDP properties 8 PCMA/8000/1
> Message: Added audio payload type to SDP properties 0 PCMU/8000/1
> Message: Added audio payload type to SDP properties 3 gsm/8000/1
> Message: Added audio payload type to SDP properties 115 lpc10-1.5/8000/1
> Message: Added audio payload type to SDP properties 111 speex_lbr-4/8000/1
> MediaStreamer-Message: dsp blocksize is 512.
>
>
> 2) After sending a call request
>
> state=0
> invite state=1
> | INFO1 | <udp.c: 191> Sending message:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Type: application/sdp
> Content-Length: 357
>
> v=0
> o=kanika 123456 654321 IN IP4 192.168.8.70
> s=A conversation
> c=IN IP4 192.168.8.70
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO1 | <udp.c: 191> Sending message:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Type: application/sdp
> Content-Length: 357
>
> v=0
> o=kanika 123456 654321 IN IP4 192.168.8.70
> s=A conversation
> c=IN IP4 192.168.8.70
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
> | INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Content-Length: 0
>
>
>
> | INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse!
>
> | INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> Content-Length: 0
>
>
>
> | INFO1 | <ict_callbacks.c: 41> OnEvent_New_Incoming1xxResponse!
>
>
> 3) On accepting request by the remote
>
> MediaStreamer-Message: dsp blocksize is 512.
> MediaStreamer-Message: ms_filter_add_link: ringplay,0 -> OssWrite,0
> MediaStreamer-Message: dsp blocksize is 512.
> | INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> Content-Type: application/sdp
> Content-Length: 302
>
> v=0
> o=saurabh 123456 654321 IN IP4 192.168.8.65
> s=A conversation
> c=IN IP4 192.168.8.65
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
>
>
> | INFO1 | <ict_callbacks.c: 71> OnEvent_New_Incoming2xxResponse!
>
> | INFO1 | <ict_callbacks.c: 122> Found body application/sdp
>
> MediaStreamer-Message: dsp blocksize is 512.
> MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> SpeexEncoder,0
> MediaStreamer-Message: ms_filter_add_link: SpeexEncoder,0 -> RTPSend,0
> MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> SpeexDecoder,0
> MediaStreamer-Message: ms_filter_add_link: SpeexDecoder,0 -> OssWrite,0
> MediaStreamer-Message: dsp blocksize is 512.
> | INFO1 | <udp.c: 191> Sending message:
> ACK sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK1250473180
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 ACK
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
> 4) On releasing connection locally
>
>
> | INFO1 | <ict_callbacks.c: 30> Transaction 1 killed.
>
> state end=2
> oRTP-stats-Message:
> Global statistics :
> packet_sent=7222
> sent=361100 bytes
> packet_recv=14447
> hw_recv=722350 bytes
> recv=270400 bytes
> unavaillable=7223 bytes
> outoftime=0
> bad=0
> discarded=181150
>
> | INFO1 | <udp.c: 191> Sending message:
> BYE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 191> Sending message:
> BYE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 160> info: RECEIVING UDP MESSAGE:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> Content-Length: 0
>
>
>
> | INFO1 | <nict_callbacks.c: 30> Transaction 2 killed.
>
> | INFO1 | <osipdialog.c: 1599> Call leg is removed. It remains 0 call legs in
> the ua list.
>
>
>
> ______________________________________________________________________
>
> Linphone 2
> -----------
>
>
> 1) Messages after starting Linphone:
>
>
> | INFO1 | <osipua.c: 59> Starting osip stack and osipua layer.
>
> MediaStreamer-Message: Detected /dev/dsp -
>
> MediaStreamer-WARNING **: Unable to find mixer for id 1.
> MediaStreamer-Message: Detected /dev/dsp1 -
>
> MediaStreamer-WARNING **: Unable to find mixer for id 3.
> | INFO1 | <udp.c: 76> Entering osipua thread.
>
> Found 2 interfaces.
> Found eth0 interface with ip address 192.168.8.65
>
> GnomeUI-WARNING **: Could not open help topics file NULL
> Adding <sip:address@hidden> to the list of alias.
> Message: Added audio payload type to SDP properties 110 speex-4/8000/1
> Message: Added audio payload type to SDP properties 0 PCMU/8000/1
> Message: Added audio payload type to SDP properties 3 gsm/8000/1
> Message: Added audio payload type to SDP properties 115 lpc10-1.5/8000/1
> Message: Added audio payload type to SDP properties 8 PCMA/8000/1
> Message: Added audio payload type to SDP properties 111 speex_lbr-4/8000/1
>
> MediaStreamer-WARNING **: dsp block size set to 1024.
> MediaStreamer-Message: dsp blocksize is 1024.
>
>
>
>
> 2) Message after receiving a call request:
>
>
> | INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Type: application/sdp
> Content-Length: 357
>
> v=0
> o=kanika 123456 654321 IN IP4 192.168.8.70
> s=A conversation
> c=IN IP4 192.168.8.70
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
>
> | INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
> INVITE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Type: application/sdp
> Content-Length: 357
>
> v=0
> o=kanika 123456 654321 IN IP4 192.168.8.70
> s=A conversation
> c=IN IP4 192.168.8.70
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111 101
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
>
>
> | INFO1 | <ist_callbacks.c: 195> OnEvent_New_IncomingInvite!
>
> | INFO1 | <ist_callbacks.c: 46> Sending 100 trying.
>
> |WARNING| <uatransaction.c: 308> ua_transaction_execute: could not get dialog
> transaction.
>
> | INFO1 | <osipua.c: 532> osip_ua_find 1: 192.168.8.65 <> 192.168.8.65
>
> | INFO1 | <osipdialog.c: 157> <sip:address@hidden>;tag=609359131 has called
> at 1043055721.
>
> | INFO1 | <ist_callbacks.c: 123> Found body application/sdp.
>
> | INFO1 | <ist_callbacks.c: 134> Creating a new body context.
>
> | INFO1 | <sdpcontext.c: 126> sdp_context_notify_inc_req: negociation
> returned: 200
>
>
> MediaStreamer-WARNING **: dsp block size set to 1024.
> MediaStreamer-Message: dsp blocksize is 1024.
> MediaStreamer-Message: ms_filter_add_link: ringplay,0 -> OssWrite,0
>
> MediaStreamer-WARNING **: dsp block size set to 1024.
> MediaStreamer-Message: dsp blocksize is 1024.
> invite handler done.
> | INFO1 | <udp.c: 191> Sending message:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Content-Length: 0
>
>
> | INFO1 | <udp.c: 191> Sending message:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> Content-Length: 0
>
>
>
> 3) On answering the call:
>
> state=3
>
> MediaStreamer-WARNING **: dsp block size set to 1024.
> MediaStreamer-Message: dsp blocksize is 1024.
> MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> SpeexEncoder,0
> MediaStreamer-Message: ms_filter_add_link: SpeexEncoder,0 -> RTPSend,0
> MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> SpeexDecoder,0
> MediaStreamer-Message: ms_filter_add_link: SpeexDecoder,0 -> OssWrite,0
>
> MediaStreamer-WARNING **: dsp block size set to 1024.
> MediaStreamer-Message: dsp blocksize is 1024.
> | INFO1 | <udp.c: 191> Sending message:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2071094665
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 INVITE
> Contact: <sip:address@hidden>
> Content-Type: application/sdp
> Content-Length: 302
>
> v=0
> o=saurabh 123456 654321 IN IP4 192.168.8.65
> s=A conversation
> c=IN IP4 192.168.8.65
> t=0 0
> m=audio 7000 RTP/AVP 110 8 0 3 115 111
> a=rtpmap:110 speex-4/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:3 gsm/8000/1
> a=rtpmap:115 lpc10-1.5/8000/1
> a=rtpmap:111 speex_lbr-4/8000/1
>
> | INFO1 | <ist_callbacks.c: 32> Transaction 1 killed.
>
> invite state=4
>
> oRTP-WARNING **: Error sending rtp packet: Connection refused.
> | INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
> ACK sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK1250473180
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 20 ACK
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
>
>
> oRTP-WARNING **: Error sending rtp packet: Connection refused.
>
> MediaStreamer-WARNING **: MSTimer: must catchup 3 ticks.
>
> MediaStreamer-WARNING **: MSTimer: must catchup 1 ticks.
>
> MediaStreamer-WARNING **: MSTimer: must catchup 1 ticks.
>
>
>
>
>
>
> 4) After disconnecting from remote:
>
>
> | INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
> BYE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
>
>
> oRTP-WARNING **: Error sending rtp packet: Connection refused.
>
> oRTP-WARNING **: Error sending rtp packet: Connection refused.
>
> oRTP-WARNING **: Error sending rtp packet: Connection refused.
> | INFO1 | <udp.c: 162> info: RECEIVING UDP MESSAGE:
> BYE sip:address@hidden SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> max-forwards: 10
> user-agent: oSIP/Linphone-0.9.1
> Content-Length: 0
>
>
>
> | INFO1 | <nist_callbacks.c: 62> nist_bye_received():
>
> | INFO1 | <osipdialog.c: 1158> call-leg has been found!
>
> oRTP-stats-Message:
> Global statistics :
> packet_sent=14662
> sent=733100 bytes
> packet_recv=7222
> hw_recv=361100 bytes
> recv=361100 bytes
> unavaillable=7325 bytes
> outoftime=0
> bad=0
> discarded=0
>
> bye handler: state=0
> | INFO1 | <udp.c: 191> Sending message:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.8.70:5060;branch=z9hG4bK2119249559
> From: <sip:address@hidden>;tag=609359131
> To: <sip:address@hidden>;tag=2119830184
> Call-ID: address@hidden
> CSeq: 21 BYE
> Content-Length: 0
>
>
>
>
> ______________________________________________________________________
>
> PC-1 lsmod output
> ------------------
>
> Module Size Used by
> via82cxxx_audio 17552 1 (autoclean)
> ac97_codec 8800 0 (autoclean) [via82cxxx_audio]
> soundcore 4464 2 (autoclean) [via82cxxx_audio]
> autofs 11264 1 (autoclean)
> 8139too 16480 1 (autoclean)
> ipchains 38976 0 (unused)
> usb-uhci 20720 0 (unused)
> usbcore 49664 1 [usb-uhci]
>
> --------------------------------------------------------------------
>
> PC-2 lsmod output
> -------------------
>
> Module Size Used by Not tainted
> via82cxxx_audio 20448 1 (autoclean)
> uart401 7936 0 (autoclean) [via82cxxx_audio]
> ac97_codec 11904 0 (autoclean) [via82cxxx_audio]
> sound 72012 0 (autoclean) [via82cxxx_audio uart401]
> soundcore 6692 4 (autoclean) [via82cxxx_audio sound]
> binfmt_misc 7556 1
> nfsd 76160 8 (autoclean)
> lockd 56736 1 (autoclean) [nfsd]
> sunrpc 75764 1 (autoclean) [nfsd lockd]
> autofs 12164 0 (autoclean) (unused)
> eepro100 20336 1
> ipchains 43560 10
> ide-cd 30272 0 (autoclean)
> cdrom 32192 0 (autoclean) [ide-cd]
> nls_iso8859-1 3488 3 (autoclean)
> nls_cp437 5120 3 (autoclean)
> vfat 12092 3 (autoclean)
> fat 37400 0 (autoclean) [vfat]
> ext3 67136 1
> jbd 49400 1 [ext3]
>
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>
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--
Simon Morlat <address@hidden>