[Asterisk-Users] SoxMix - Fails to Execute

Chris Glover chris at glovercc.clara.co.uk
Thu Jul 15 14:48:51 MST 2004


Is the path to soxmix in the $PATH environment variable when asterisk
starts. If you're running from an init script it may not have path set at
that point.

When you log in, you set the path variable. Have you tried putting
explicit paths into the command in your extensions.conf?

IE /usr/bin/soxmix instead of just soxmix.

HTH

Chris

-- 
Chris
----------------------------------
E Mail:	chris at glovercc.clara.co.uk
SIP: 84411389 at voiptalk.org
IAXTEL: 17003366726

On Thu, 15 Jul 2004, Chris Smales - Magenta Solutions wrote:

> I have Asterisk configured to record calls. Both in and out record ok
> but SoxMix fails to join the two files.
> The error from the CLI is as follows:
>
> Execute of ( nice -n 19 soxmix
> /var/spool/asterisk/monitor/Support-in.wav
> /var/spool/asterisk/monitor/Support-out.wav
> /var/spool/asterisk/monitor/Support.wav && rm -f
> /var/spool/asterisk/monitor/Support-* ) & failed.
>
> If I run exactly the same command from Linux it runs ok and the two
> files get mixed.
>
> Can anybody suggest why Asterisk has a problem running the command?
>
> Thanks,
> Chris
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> asterisk-users-request at lists.digium.com
> Sent: 15 July 2004 21:50
> To: asterisk-users at lists.digium.com
> Subject: Asterisk-Users digest, Vol 1 #4559 - 12 msgs
>
> Send Asterisk-Users mailing list submissions to
> 	asterisk-users at lists.digium.com
>
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> or, via email, send a message with subject or body 'help' to
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> "Re: Contents of Asterisk-Users digest..."
>
>
> Today's Topics:
>
>    1. Re: Bounty!  For help with echo cancellation code.
> (echo-asterisk at secondphone.com)
>    2. RE: Updated Grandstream configurator (Mike Reed)
>    3. RE: Re: VoicePulse changes (Mike Reed)
>    4. freenode #asterisk IRC channel identd problem (Nathan Alpert)
>    5. Re: freenode #asterisk IRC channel identd problem (Olle E.
> Johansson)
>    6. Re: Re: Problem loadin oh323 solved (ruixun wu)
>    7. bristuff 0.0.3 ? (Bjoern Adler)
>    8. RE: VoicePulse changes (daryl at introspect.net)
>    9. RE: freenode #asterisk IRC channel identd problem (Mike Reed)
>   10. Re: freenode #asterisk IRC channel identd problem (Steven
> Critchfield)
>   11. Re: Updated Grandstream configurator (Stephen R. Besch)
>   12. Re: Updated Grandstream configurator (Stephen R. Besch)
>
> --__--__--
>
> Message: 1
> Date: Thu, 15 Jul 2004 12:39:45 -0700
> From: echo-asterisk at secondphone.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Bounty!  For help with echo cancellation
> code.
> Reply-To: asterisk-users at lists.digium.com
>
> On Wed, Jul 14, 2004 at 10:14:19AM -0700, Bob Knight wrote:
> > echo-asterisk at secondphone.com wrote:
> > >>>From the CLI and during a call I want to be able to:
> > >
> > >  *** Pulse the outgoing line and record at least 50 ms of the
> > > incoming  line.
> > >
> > >      The pulse waveform must be specifiable as a series of
> amplitudes
> > >      for each 1/8000 sec time slot.  It would be best of these
> values
> > >      could be read from a file specified on the CLI command line.
> > >
> > >      Timing should be synced between the pulse and the echo so that
> the
> > >      delay from the pulse to the echo can be accurately determined.
> > >
> > >      Echo cancellation should be disabled during this operation.
> > >
> > >      This would operate similar to the echo-training code that
> operates
> > >      at the initiation of a call except that this could be done at
> > >      any time.
> > >
> > >      The initial pulse and any echoes can be combined and saved in a
> > >      single channel.
> > >
> > >      Output should go to a file and should be in a simple format
> that
> > >      a program such as Audacity can read, display and play.
> > >
> > >
> > >  *** Pulse the outgoing line and record at least 50 ms of the
> > > incoming  line.
> > >
> > >      Same as above EXCEPT echo cancellation would not be disabled
> during
> > >      this test and the results of the echo cancellation operations
> should
> > >      be recorded and saved in a separate channel.
> > >
> > >
> > >  *** Change variables used to control echo cancellation.
> > >
> > >      Only the code in mec2.h is of interest.
> > >
> > >      I will help identify the variables and modify the mec2.h code
> as
> > >      needed to accomplish this goal.
> > >
> > >      There are a lot of parameters in mec2.h that may affect the
> quality
> > >      of the echo cancellation.  I want to be able to adjust them 'on
> the
> > >      fly' and be able to immediately hear the results.
> > >
> > >
> > >I am open to alternative proposals which would accomplish the same
> goals.
> > >
> > >Name your price.
> >
> > How about being able to "see" the results real time?
> > I use a package called SMAART from siasoft.com.
> > It is a dual channel spectrum analyzer.
> > Run the output line as your reference channel and the input line as
> > your measurement channel.
> >
> > You can get great info from the impulse response and transfer
> > function.
> >
> > You could also use this to compare different codecs.
> > The impulse function will tell you how long it takes.
> > The transfer function will tell you just how good a job it did at
> > reconstruction the original audio.
> >
>
> Almost 20 years ago I wrote my own digital spectrum analyzer code which
> I then used to do my research.  Provided that SMAART can fully utilize
> the transfer function (do convolutions etc) it would may be useful, but
> spectrum analysis is not the hard part.  Controlling and getting the
> data out of zaptel.o is the hard part and help with that is what is
> requested in the Bounty!
>
> echo
>
>
>
> > --
> > Bob Knight
> > [-w] the work option
> > bk at minusw.com
> > 925-449-9163
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --__--__--
>
> Message: 2
> Subject: RE: [Asterisk-Users] Updated Grandstream configurator
> Date: Thu, 15 Jul 2004 14:48:10 -0500
> From: "Mike Reed" <mike.reed at voxpath.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> =20
>
> > -----Original Message-----
> > The bad part is that starting with SP2 on w2k ms EULA has=20  changed
> >to include=20  your agreement to let microsoft not only see, what you
> >have=20  on your computer,=20  but also install software on it. This
> >has caused a big=20  corporate hold on=20  updating beyond SP2. The
> >medical industry in particular is=20  having a hard=20  time, as ms has
>
> >not signed a non disclosure to have access to=20  personal=20  medical
> >information.
> >=20
> > - --=20
> > Steve
>
> This simply isn't true.  The medical law you're referring to is HIPPA,
> and there's *nothing* in the Microsoft EULA that allows them to read
> otherwise proprietary or personal information off your system.
>
> While I'm not a Microsoft apologist, I can stand to see them slammed
> from purely ignorant disinformation.
>
> Mike :)
>
> --__--__--
>
> Message: 3
> Subject: RE: [Asterisk-Users] Re: VoicePulse changes
> Date: Thu, 15 Jul 2004 14:50:01 -0500
> From: "Mike Reed" <mike.reed at voxpath.com>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> +3, Funny=20
>
> > -----Original Message-----
> > Maybe if you circle the globe enough times, crossing the=20
> >international=20  date line each time, of course, it would be possible
> >to get to August=20  15th yesterday  ;-) =20  SRB
>
>
> --__--__--
>
> Message: 4
> From: Nathan Alpert <asterisk at demicrosystems.com>
> To: asterisk-users at lists.digium.com
> Organization: Digital Evolution Microsystems
> Date: 15 Jul 2004 14:54:01 -0500
> Subject: [Asterisk-Users] freenode #asterisk IRC channel identd problem
> Reply-To: asterisk-users at lists.digium.com
>
> Sorry to ask this question here since it's related to IRC and not
> Asterisk, but I am having trouble logging into the #asterisk IRC channel
> on freenode and was wondering if anyone else has had this problem and
> solved it.
>
> So here's the situation: Whenever I try to login to the #asterisk
> channel I get a message like "you must be identified to login to this
> channel." So after doing a little research on this, I found that the
> identity thing is related to the "identd" server used to identify
> computers on a network. Apparently in Linux there is some identd server
> thing that needs to be configured so I didn't mess with this and used
> mIRC on Windows which has a identd server built into the program. I have
> my firewall forwarding port 113 to the computer running mIRC (which is
> apparently needed to listen for the identd requests) and the SOB still
> gives me the same message and I can't get into the #asterisk channel.
>
> Has anyone else had this problem and solved it?
>
> Thanks,
> Nate Alpert
> asterisk at demicrosystems.com
>
>
> --__--__--
>
> Message: 5
> Date: Thu, 15 Jul 2004 22:07:36 +0200
> From: "Olle E. Johansson" <oej at edvina.net>
> Organization: Edvina AB
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] freenode #asterisk IRC channel identd
> problem
> Reply-To: asterisk-users at lists.digium.com
>
> Nathan Alpert wrote:
>
> > Sorry to ask this question here since it's related to IRC and not
> > Asterisk, but I am having trouble logging into the #asterisk IRC
> channel
> > on freenode and was wondering if anyone else has had this problem and
> > solved it.
> >
> > So here's the situation: Whenever I try to login to the #asterisk
> > channel I get a message like "you must be identified to login to this
> > channel." So after doing a little research on this, I found that the
> > identity thing is related to the "identd" server used to identify
> Please read the following helptext from Asterisk.org:
>
> "The Asterisk channel now requires that your nick be registered with the
> Freenode Nickerv in order to participate. This measure has been
> taken to combat spambots and the like. We apologize for the
> inconvenience. Please "/msg NickServ help register" in your IRC client
> to learn
> how to register your nick"
>
> This has nothing to do with identd. Run the command and you'll get
> assistance.
>
> /O
>
> --__--__--
>
> Message: 6
> Date: Thu, 15 Jul 2004 16:10:12 -0400 (EDT)
> From: ruixun wu <ruixunwu at yahoo.ca>
> Subject: Re: [Asterisk-Users] Re: Problem loadin oh323 solved
> To: asterisk-users at lists.digium.com
> Reply-To: asterisk-users at lists.digium.com
>
> Hi,
>    Thanks for you reply.
>    I download glibc-2.3.2.tar.gz and
> glibc-linuxthreads-2.3.2.tar.gz. The configuration
> process was fine, no error occured(I typed
> glibc-2.3.2/configure --enable-add-ons=linuxthreads).
> But there was a strange things happened in make
> process. The make process kept compling the files
> located in directory glibc-2.3.2/csu and wouldn't
> stop. At first I didn't notice this and waiting for 4
> hours.
>
>    Do you have any idea?
>
> Thanks a lot
> Rui
>
> --- Lars Degenhardt <lars at lcc-degenhardt.de> wrote: >
> ruixun wu wrote:
> > > Hello Soumaya,
> > >
> > >    It's great that you solved the problem.
> > >    But I still don't know how to do. What's the
> > > problem with redhat 9.0? Could you tell me more
> > > details?
> > >
> > > Thanks a lot
> > > Rui
> > >
> >
> > as I am the "kind member" I can tell you also:
> >
> > get the latest glibc and libssl updates and
> > recompile the whole bunch
> > (pwlib/opneh323/asterisk-oh323)
> >
> > >
> > > Fathallah Soumaya wrote:
> > >
> > >>Hello everybody,
> > >>
> > >>The problem that I had withj loading oh323 module
> > >
> > > was finally solved
> > >
> > >>thanks to the help of a kind member of this list,
> > it
> > >
> > > was due to a
> > >
> >
> > --
> > Lars Degenhardt
> > phon: +49 76814749263| mobile: +49 1736936968| box:
> > +49 891488262647
> > BOFH excuse #83:
> > Support staff hung over, send aspirin and come back
> > LATER.
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ______________________________________________________________________
> Post your free ad now! http://personals.yahoo.ca
>
> --__--__--
>
> Message: 7
> From: "Bjoern Adler" <asterisk at brainhawk.de>
> To: <asterisk-users at lists.digium.com>
> Date: Thu, 15 Jul 2004 22:19:25 +0200
> Subject: [Asterisk-Users] bristuff 0.0.3 ?
> Reply-To: asterisk-users at lists.digium.com
>
> Hi all,
>
> are there any news about bristuff 0.0.3, which compiles against CVS =
> HEAD?
>
> Any informations regarding the timeframe of appearance would be =
> appreciated...
>
> Greetings
>
> Bjoern
>
>
>
>
> --__--__--
>
> Message: 8
> Subject: RE: [Asterisk-Users] VoicePulse changes
> Date: Thu, 15 Jul 2004 16:21:42 -0400
> From: <daryl at introspect.net>
> To: <asterisk-users at lists.digium.com>
> Reply-To: asterisk-users at lists.digium.com
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com=20
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Jay Milk
> > Sent: Thursday, July 15, 2004 1:00 PM
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] VoicePulse changes
> >=20
> > Welcome back to July.  How's the future?
> >=20
> > There's one rather reliable, albeit not very popular provider=20
> > with DIDs in the Philly area:  Vonage.  Their softphone=20
>
> [...]
>
> I, as well as almost everyone else on this list, is very well aware of
> Vonage.  As soon as they start officially supporting Asterisk and
> specify things like whether you can have concurrent inbound calls
> without additional charge, calls rolling over, etc it just might be a
> viable option.
>
> No, my Asterisk installation is not in my basement being used as a
> glorified answering machine.  People who use these things for actual
> business systems need more than "I played around with <x> and got <y> to
> work!  Cool!  Maybe it will even keep working if <x> doesn't decide to
> change it or start charging me, etc."
>
> For now, that leaves people in my position paying for PRIs or POTS lines
> just to be sure.
>
> Daryl
>
> --__--__--
>
> Message: 9
> Subject: RE: [Asterisk-Users] freenode #asterisk IRC channel identd
> problem
> Date: Thu, 15 Jul 2004 15:31:07 -0500
> From: "Mike Reed" <mike.reed at voxpath.com>
> To: <asterisk-users at lists.digium.com>
> Cc: <asterisk at demicrosystems.com>
> Reply-To: asterisk-users at lists.digium.com
>
> It's got nothing to do with IdentD and everything to do with registering
> your nick on the net/node.
>
> Mike ;)=20
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com=20
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of=20
> > Nathan Alpert
> > Sent: Thursday, July 15, 2004 2:54 PM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] freenode #asterisk IRC channel=20
> > identd problem
> >=20
> > Sorry to ask this question here since it's related to IRC and not
> > Asterisk, but I am having trouble logging into the #asterisk=20
> > IRC channel
> > on freenode and was wondering if anyone else has had this problem and
> > solved it.
> >=20
> > So here's the situation: Whenever I try to login to the #asterisk
> > channel I get a message like "you must be identified to login to this
> > channel." So after doing a little research on this, I found that the
> > identity thing is related to the "identd" server used to identify
> > computers on a network. Apparently in Linux there is some=20
> > identd server
> > thing that needs to be configured so I didn't mess with this and used
> > mIRC on Windows which has a identd server built into the=20
> > program. I have
> > my firewall forwarding port 113 to the computer running mIRC (which is
> > apparently needed to listen for the identd requests) and the SOB still
> > gives me the same message and I can't get into the #asterisk channel.
> >=20
> > Has anyone else had this problem and solved it?=20
> >=20
> > Thanks,
> > Nate Alpert
> > asterisk at demicrosystems.com
> >=20
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >=20
> >=20
>
> --__--__--
>
> Message: 10
> Subject: Re: [Asterisk-Users] freenode #asterisk IRC channel identd
> problem
> From: Steven Critchfield <critch at basesys.com>
> To: asterisk-users at lists.digium.com
> Date: Thu, 15 Jul 2004 15:31:35 -0500
> Reply-To: asterisk-users at lists.digium.com
>
> On Thu, 2004-07-15 at 14:54, Nathan Alpert wrote:
> > Sorry to ask this question here since it's related to IRC and not
> > Asterisk, but I am having trouble logging into the #asterisk IRC
> channel
> > on freenode and was wondering if anyone else has had this problem and
> > solved it.
> >
> > So here's the situation: Whenever I try to login to the #asterisk
> > channel I get a message like "you must be identified to login to this
> > channel." So after doing a little research on this, I found that the
> > identity thing is related to the "identd" server used to identify
> > computers on a network. Apparently in Linux there is some identd
> server
> > thing that needs to be configured so I didn't mess with this and used
> > mIRC on Windows which has a identd server built into the program. I
> have
> > my firewall forwarding port 113 to the computer running mIRC (which is
> > apparently needed to listen for the identd requests) and the SOB still
> > gives me the same message and I can't get into the #asterisk channel.
> >
> > Has anyone else had this problem and solved it?
>
> search THIS list to find that you must be registered to nickserv to get
> in the channel. It has been discussed many times and at length.
>



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