Chris Smales - Magenta Solutions
2004-Jul-15 14:15 UTC
[Asterisk-Users] SoxMix - Fails to Execute
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@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: 15 July 2004 21:50 To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #4559 - 12 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Bounty! For help with echo cancellation code. (echo-asterisk@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@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@secondphone.com To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Bounty! For help with echo cancellation code. Reply-To: asterisk-users@lists.digium.com On Wed, Jul 14, 2004 at 10:14:19AM -0700, Bob Knight wrote:> echo-asterisk@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 ofamplitudes> > for each 1/8000 sec time slot. It would be best of thesevalues> > could be read from a file specified on the CLI command line. > > > > Timing should be synced between the pulse and the echo so thatthe> > 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 thatoperates> > 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 formatthat> > 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 disabledduring> > this test and the results of the echo cancellation operationsshould> > 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 codeas> > needed to accomplish this goal. > > > > There are a lot of parameters in mec2.h that may affect thequality> > of the echo cancellation. I want to be able to adjust them 'onthe> > fly' and be able to immediately hear the results. > > > > > >I am open to alternative proposals which would accomplish the samegoals.> > > >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@minusw.com > 925-449-9163 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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@voxpath.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@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 > SteveThis 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@voxpath.com> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@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@demicrosystems.com> To: asterisk-users@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@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@demicrosystems.com --__--__-- Message: 5 Date: Thu, 15 Jul 2004 22:07:36 +0200 From: "Olle E. Johansson" <oej@edvina.net> Organization: Edvina AB To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] freenode #asterisk IRC channel identd problem Reply-To: asterisk-users@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 IRCchannel> 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 identifyPlease 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@yahoo.ca> Subject: Re: [Asterisk-Users] Re: Problem loadin oh323 solved To: asterisk-users@lists.digium.com Reply-To: asterisk-users@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@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@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@brainhawk.de> To: <asterisk-users@lists.digium.com> Date: Thu, 15 Jul 2004 22:19:25 +0200 Subject: [Asterisk-Users] bristuff 0.0.3 ? Reply-To: asterisk-users@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@introspect.net> To: <asterisk-users@lists.digium.com> Reply-To: asterisk-users@lists.digium.com> -----Original Message----- > From: asterisk-users-admin@lists.digium.com=20 > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk > Sent: Thursday, July 15, 2004 1:00 PM > To: asterisk-users@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@voxpath.com> To: <asterisk-users@lists.digium.com> Cc: <asterisk@demicrosystems.com> Reply-To: asterisk-users@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@lists.digium.com=20 > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of=20 > Nathan Alpert > Sent: Thursday, July 15, 2004 2:54 PM > To: asterisk-users@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@demicrosystems.com >=20 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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@basesys.com> To: asterisk-users@lists.digium.com Date: Thu, 15 Jul 2004 15:31:35 -0500 Reply-To: asterisk-users@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 IRCchannel> 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 identdserver> 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. Ihave> 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. -- Steven Critchfield <critch@basesys.com> --__--__-- Message: 11 To: asterisk-users@lists.digium.com From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu> Date: Thu, 15 Jul 2004 16:31:47 -0400 Subject: [Asterisk-Users] Re: Updated Grandstream configurator Reply-To: asterisk-users@lists.digium.com Steve wrote:> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote: > >>The most recent version of GSConfigure is available at >>www.buffalo.edu/~sbesch Several serious bugs that kept the programfrom>>getting started have been ferreted out and corrected with the help of >>Bruce Komito. The program is now actually running on someone's machine >>other than mine. I have built this version with the oldest copies ofthe>>system dll's that I could find inn an effort to solve the VB setupbug,>>so, hopefully it will no longer send anyone through multiple restarts. >>You should have at least SP3, or even better, SP4 on Win2k. I believeit>>will run on Win9x, but I have not tested it and can make noguarantees.>> >>Steve Besch > > > The bad part is that starting with SP2 on w2k ms EULA has changed toinclude> your agreement to let microsoft not only see, what you have on yourcomputer,> but also install software on it. This has caused a big corporate holdon> updating beyond SP2. The medical industry in particular is having ahard> time, as ms has not signed a non disclosure to have access to personal> medical information. > > - -- > Steve > > "They that would give up essential liberty for temporary safetydeserve> neither liberty nor safety." > Benjamin Franklin >Since I am quite sure that the program will run without updating any of the dll's, what I should do is simply register them with regsvr32 from a batch job and bag the VB6 installer altogether. Before I do that though, can anyone tell me if regsvr32 ships with standard Win2k/WinXP? Stephen R. Besch --__--__-- Message: 12 Date: Thu, 15 Jul 2004 16:31:47 -0400 From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Updated Grandstream configurator Reply-To: asterisk-users@lists.digium.com Steve wrote:> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote: > >>The most recent version of GSConfigure is available at >>www.buffalo.edu/~sbesch Several serious bugs that kept the programfrom>>getting started have been ferreted out and corrected with the help of >>Bruce Komito. The program is now actually running on someone's machine >>other than mine. I have built this version with the oldest copies ofthe>>system dll's that I could find inn an effort to solve the VB setupbug,>>so, hopefully it will no longer send anyone through multiple restarts. >>You should have at least SP3, or even better, SP4 on Win2k. I believeit>>will run on Win9x, but I have not tested it and can make noguarantees.>> >>Steve Besch > > > The bad part is that starting with SP2 on w2k ms EULA has changed toinclude> your agreement to let microsoft not only see, what you have on yourcomputer,> but also install software on it. This has caused a big corporate holdon> updating beyond SP2. The medical industry in particular is having ahard> time, as ms has not signed a non disclosure to have access to personal> medical information. > > - -- > Steve > > "They that would give up essential liberty for temporary safetydeserve> neither liberty nor safety." > Benjamin Franklin >Since I am quite sure that the program will run without updating any of the dll's, what I should do is simply register them with regsvr32 from a batch job and bag the VB6 installer altogether. Before I do that though, can anyone tell me if regsvr32 ships with standard Win2k/WinXP? Stephen R. Besch --__--__-- _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest
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@glovercc.clara.co.uk SIP: 84411389@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@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > asterisk-users-request@lists.digium.com > Sent: 15 July 2004 21:50 > To: asterisk-users@lists.digium.com > Subject: Asterisk-Users digest, Vol 1 #4559 - 12 msgs > > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific than > "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Bounty! For help with echo cancellation code. > (echo-asterisk@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@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@secondphone.com > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Bounty! For help with echo cancellation > code. > Reply-To: asterisk-users@lists.digium.com > > On Wed, Jul 14, 2004 at 10:14:19AM -0700, Bob Knight wrote: > > echo-asterisk@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@minusw.com > > 925-449-9163 > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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@voxpath.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@voxpath.com> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@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@demicrosystems.com> > To: asterisk-users@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@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@demicrosystems.com > > > --__--__-- > > Message: 5 > Date: Thu, 15 Jul 2004 22:07:36 +0200 > From: "Olle E. Johansson" <oej@edvina.net> > Organization: Edvina AB > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] freenode #asterisk IRC channel identd > problem > Reply-To: asterisk-users@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@yahoo.ca> > Subject: Re: [Asterisk-Users] Re: Problem loadin oh323 solved > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@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@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@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@brainhawk.de> > To: <asterisk-users@lists.digium.com> > Date: Thu, 15 Jul 2004 22:19:25 +0200 > Subject: [Asterisk-Users] bristuff 0.0.3 ? > Reply-To: asterisk-users@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@introspect.net> > To: <asterisk-users@lists.digium.com> > Reply-To: asterisk-users@lists.digium.com > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com=20 > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk > > Sent: Thursday, July 15, 2004 1:00 PM > > To: asterisk-users@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@voxpath.com> > To: <asterisk-users@lists.digium.com> > Cc: <asterisk@demicrosystems.com> > Reply-To: asterisk-users@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@lists.digium.com=20 > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of=20 > > Nathan Alpert > > Sent: Thursday, July 15, 2004 2:54 PM > > To: asterisk-users@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@demicrosystems.com > >=20 > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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@basesys.com> > To: asterisk-users@lists.digium.com > Date: Thu, 15 Jul 2004 15:31:35 -0500 > Reply-To: asterisk-users@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. >
Any help setting up a X101P in Spain zttool show it as UNCONFIGURED (or in RED when line is out, so the card is running ok) zaptel.conf loadzone = fr defaultzone = fr fxsks=1 zapata.conf ; ; Zapata telephony interface sample configuration file ; [channels] ; ; X100P plugged into PSTN ; context=incoming signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel => 1 Adri? Vidal