I've been doing everything the forum member's been suggesting me, I even attached my conf files as well. But I haven't gotten anywhere that's the only reason I'm keep on replying to these emails...sorry<br><br><br><b><i>asterisk-dev-request@lists.digium.com</i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> Send asterisk-dev mailing list submissions to<br> asterisk-dev@lists.digium.com<br><br>To subscribe or unsubscribe via the World Wide Web, visit<br> http://lists.digium.com/mailman/listinfo/asterisk-dev<br>or, via email, send a message with subject or body 'help' to<br> asterisk-dev-request@lists.digium.com<br><br>You can reach the person managing the list at<br> asterisk-dev-owner@lists.digium.com<br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of asterisk-dev digest..."<br><br><br>Today's Topics:<br><br> 1. Re: plese help (Russell Bryant)<br>
2. Re: plese help (Matthew Rubenstein)<br> 3. (Update) Digium FTP server will be replaced with HTTP server<br> (Kevin P. Fleming)<br> 4. Re: asterisk-dev Digest, Vol 36, Issue 68 (jazzy singh)<br> 5. Re: asterisk broke and I'm getting fired from<br> (Eric "ManxPower" Wieling)<br> 6. Re: asterisk-dev Digest, Vol 36, Issue 68 (Russell Bryant)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Thu, 26 Jul 2007 17:51:43 -0400<br>From: Russell Bryant <russell@digium.com><br>Subject: Re: [asterisk-dev] plese help<br>To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com><br>Message-ID: <46A9176F.1090400@digium.com><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>jazzy singh wrote:<br>> Please help......I'm using asterisk-1.2.13 with lumenvox, festival.. I <br>> was able to compile and install it without any problem,,as soon as I run <br>> asterisk it
dies without giving any kind of error. Anybody please help ....<br><br>You have got to be kidding me ... You have now started 4 threads asking for <br>help on the same thing. You have received many suggestions for things to check <br>for, and you have not responded to most of them.<br><br>This is not intended to be a technical support forum. Please try the <br>asterisk-users mailing list to get help. If you are able to determine a problem <br>*with an up to date version*, then you can file a bug on bugs.digium.com. We <br>don't handle bug reports through this mailing list so that the information stays <br>organized and doesn't get lost.<br><br>-- <br>Russell Bryant<br>Software Engineer<br>Digium, Inc.<br><br><br><br>------------------------------<br><br>Message: 2<br>Date: Thu, 26 Jul 2007 17:55:28 -0400<br>From: Matthew Rubenstein <email@mattruby.com><br>Subject: Re: [asterisk-dev] plese help<br>To: David Boyd <dboyd@ignitetrx.com><br>Cc: Asterisk -Dev
<asterisk-dev@lists.digium.com><br>Message-ID: <1185486928.21464.153.camel@pont><br>Content-Type: text/plain<br><br> If he can't even ask for Asterisk help properly, including find the<br>right support list (-users), even after told to do so, then maybe he<br>should get fired for leaving his boss' phone system down for over a<br>whole business day. Not everyone is cut out to do this themselves - some<br>people need to hire consultants to help them, or at least to train them,<br>or to do both. Like how to plan for inevitable outages.<br><br><br>On Thu, 2007-07-26 at 16:49 -0500, asterisk-dev-request@lists.digium.com<br>wrote:<br>> Date: Thu, 26 Jul 2007 17:44:51 -0400<br>> From: David Boyd <dboyd@ignitetrx.com><br>> Subject: Re: [asterisk-dev] plese help<br>> To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com><br>> Message-ID: <1185486292.9441.45.camel@d9100><br>> Content-Type: text/plain<br>> <br>> Hi Jazzy,<br>>
<br>> Why don't you help yourself and post configs, versions, changes that<br>> you<br>> made to the system, copies of error and logs etc. and someone might<br>> help<br>> you. However I am of the opinion that you don't really want assistance<br>> you simply are having fun screwing with everybody else. <br>> <br>> Next point the application can't really die without any errors at<br>> all. <br>> There has to be some indication of what took place, so I go back to my<br>> original statement that you are just screwing with mine and everybody<br>> else' time.<br>> <br>> <br>> <br>> Dave <br>-- <br><br>(C) Matthew Rubenstein<br><br><br><br><br>------------------------------<br><br>Message: 3<br>Date: Thu, 26 Jul 2007 17:17:29 -0500<br>From: "Kevin P. Fleming" <kpfleming@digium.com><br>Subject: [asterisk-dev] (Update) Digium FTP server will be replaced<br> with HTTP server<br>To: Asterisk Developers Mailing List
<asterisk-dev@lists.digium.com><br>Message-ID: <46A91D79.7000807@digium.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>Some time in the next two weeks, Digium will be shutting down our FTP<br>server, located at ftp.digium.com, and begin using only the existing<br>HTTP server on the same system instead.<br><br>We have decided to only offer our public downloads over the HTTP<br>protocol, not the FTP protocol, primarily for reasons related to our<br>marketing department :-)<br><br>The site has a new name, downloads.digium.com, and will also respond to<br>the old name ftp.digium.com, but will no longer respond to requests made<br>via the FTP protocol; only the HTTP protocol will be supported. There<br>should be no other user-visible changes when this change is made to the<br>server. We will begin using the downloads.digium.com name in all our<br>announcements, security advisories, documentation and other places<br>immediately, and encourage the community to
do so as well.<br><br>-- <br>Kevin P. Fleming<br>Director of Software Technologies<br>Digium, Inc. - "The Genuine Asterisk Experience" (TM)<br><br><br><br><br><br>------------------------------<br><br>Message: 4<br>Date: Thu, 26 Jul 2007 15:32:49 -0700 (PDT)<br>From: jazzy singh <jazzy_gill@yahoo.com><br>Subject: Re: [asterisk-dev] asterisk-dev Digest, Vol 36, Issue 68<br>To: asterisk-dev@lists.digium.com<br>Message-ID: <465433.11341.qm@web50703.mail.re2.yahoo.com><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>Hey David,<br> I apologize if you thought that I'm wasting everyone's time overhere ...I'm not...I'm under a lot of frustration right now....I just want this thing to start working like it was till yesterday. I'm a newbie at asterisk so I'm discovering things as I go I figure out how to make it print error messages when i restart here'we go<br><br>version = I'm using asterisk-1.2.13 (it worked fine till last nite..when I was testing it and
must've called 20, 30)<br>my version of zaptel -1.2.10<br>lumenvox7.5<br><br>this is the error I get when starting asterisk<br><br>Asterisk Dynamic Loader Starting:<br> == Parsing '/etc/asterisk/modules.conf': Found<br> [res_musiconhold.so] => (Music On Hold Resource)<br> == Registered application 'MusicOnHold'<br> == Registered application 'WaitMusicOnHold'<br> == Registered application 'SetMusicOnHold'<br> == Registered application 'StartMusicOnHold'<br> == Registered application 'StopMusicOnHold'<br> == Parsing '/etc/asterisk/musiconhold.conf': Found<br> [res_speech.so] => (Lumenvox Speech Recognition)<br>Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:758 load_module: Lumenvox SRE Connector module Copyright (C) 1999-2007 Digium, Inc.<br>Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:759 load_module: This module is supplied under a commercial license granted by Digium, Inc.<br> == Parsing '/etc/asterisk/lumenvox.conf': Found<br> -- Using
server(s): 127.0.0.1<br> -- Loaded grammar 'plan'<br> -- Loaded grammar 'one_to_ten'<br> -- Loaded tweaking profile default<br>asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_speech.so: undefined symbol: ast_speech_register<br><br><br><br><br><br>this is what my modules.conf looks like...<br>;<br>; Asterisk configuration file<br>;<br>; Module Loader configuration file<br>;<br><br>[modules]<br>autoload=yes<br>;<br>; Any modules that need to be loaded before the Asterisk core has been<br>; initialized (just after the logger has been initialized) can be loaded<br>; using 'preload'. This will frequently be needed if you wish to map all<br>; module configuration files into Realtime storage, since the Realtime<br>; driver will need to be loaded before the modules using those configuration<br>; files are initialized.<br>;<br>; An example of loading ODBC support would be:<br>;preload => res_odbc.so<br>;preload => res_config_odbc.so<br>;<br>; If you
want, load the GTK console right away.<br>; Don't load the KDE console since<br>; it's not as sophisticated right now.<br>;<br>noload => pbx_gtkconsole.so<br>;load => pbx_gtkconsole.so<br>noload => pbx_kdeconsole.so<br>;<br>; Intercom application is obsoleted by<br>; chan_oss. Don't load it.<br>;<br>noload => app_intercom.so<br>;<br>; The 'modem' channel driver and its subdrivers are<br>; obsolete, don't load them.<br>;<br>noload => chan_modem.so<br>noload => chan_modem_aopen.so<br>noload => chan_modem_bestdata.so<br>noload => chan_modem_i4l.so<br>;<br>load => res_musiconhold.so<br>;<br>; Load either OSS or ALSA, not both<br>; By default, load OSS only (automatically) and do not load ALSA<br>;<br>noload => chan_alsa.so<br>load => res_speech.so<br>load => res_speech_lumenvox.so<br>;noload => chan_oss.so<br>;<br>; Module names listed in "global" section will have symbols globally<br>; exported to modules loaded after
them.<br>;<br>[global]<br><br><br><br><br>this is what my lumenvox.com looks like <br><br>; Lumenvox configuration file<br><br>[general]<br>servers=127.0.0.1 ; SRE Servers to use<br>save_sound_files=no ; Set to yes to save sound files for tuner usage<br><br>; Pre-loaded grammars<br>[grammars]<br>plan=/etc/asterisk/grammars/plan.gram<br>one_to_ten=/etc/asterisk/grammars/one_to_ten.gram<br><br>; Lumenvox tweaking profiles<br>; A tweaking profile can be used by using the SPEECH_ENGINE dialplan function. For example,<br>; to apply the default profile you would use Set(SPEECH_ENGINE(profile)=default)<br><br>; NOTE: Each option can also be set in the dialplan by using Set(SPEECH_ENGINE(name)=value)<br>; Example: Set(SPEECH_ENGINE(vad_eos_delay)=100)<br><br>; Default settings<br>[default]<br>; SNR threshold/Barge-In Level.<br>; An audio frame will be considered for voice activity only when the SNR metric is higher than this threshold.<br>; So for noisy channel, this value should
be lower, so that it is easier to barge in.<br>; NOTE: this value is not a measurement in dB. It is just a relative value compared to an internal standard.<br>vad_bargein_level=40<br><br>; End-of-speech delay in ms.<br>; The initial audio used to initialize voice activity detector.<br>; Setting this parameter to 60 means using the first 60ms as initialization data.<br>vad_eos_delay=2000<br><br>; Noise floor threshold.<br>; An audio frame will be considered for voice activity only when the average energy<br>; is higher than this threshold. The default value is 200. This parameter is<br>; particularly useful when the echo canceler doesn't work very well.<br>; Because the residual echo has all features of voice, so the only way we<br>; can filter them out is using a hard constraint on energy.<br>vad_noise_floor=200<br><br>; Wind back.<br>; The length of audio wound back from the point of voice detection.<br>; It helps in the situation of weak speech onset. The resolution of
this<br>; parameter is 1/8 sec,i.e. 125ms, which means setting this value to 249ms<br>; is same as setting it to 125ms.<br>vad_wind_back=255<br><br>; Burst control threshold in ms.<br>; Barge-in will be triggered only when the duration of voice is longer<br>; than this threshold.<br>vad_burst_threshold=100<br><br>; After barge-in, the streaming interface will flag that speech was detected<br>; if it detected it in the time frame specified by this option.<br>end_of_speech_timeout=8000<br><br>; Whether to use the out-of-vacabulary filter during decode.<br>use_oov_filter=no<br><br><br> <br>---------------------------------<br>Be a better Globetrotter. Get better travel answers from someone who knows.<br>Yahoo! Answers - Check it out.<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20070726/ae2167d4/attachment-0001.htm <br><br>------------------------------<br><br>Message:
5<br>Date: Thu, 26 Jul 2007 17:55:05 -0500<br>From: "Eric \"ManxPower\" Wieling" <eric@fnords.org><br>Subject: Re: [asterisk-dev] asterisk broke and I'm getting fired from<br>To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com><br>Message-ID: <46A92649.1080305@fnords.org><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>Start Asterisk as "asterisk -cvvv"<br><br><br><br><br>jazzy singh wrote:<br>> I did that but it still is dying,,,,just to let you guys know I'm using lumenvox as well, when I restart asterisk it restarts and then it just dies right away without any error message. Please help<br>> Thanks<br>> <br>> <br>> <br>> <br>> Date: Thu, 26 Jul 2007 14:20:59 -0400<br>> From: "harish kasiviswanathan" <harish.kasiviswanathan@gmail.com><br>> Subject: Re: [asterisk-dev] asterisk broke and I'm getting fired from<br>> my job :(....<br>> To: "Asterisk Developers Mailing List"
<asterisk-dev@lists.digium.com><br>> Message-ID:<br>> <8030b59a0707261120u61afd7dax72aa99be40c3e9f0@mail.gmail.com><br>> Content-Type: text/plain; charset="iso-8859-13"<br>> <br>> Looks like you are using the old version of zaptel driver and new version of<br>> asterisk. Upgrade to zaptel-1.4.4 or newer and this error should go away.<br>> <br>> harish<br>> <br>> On 7/26/07, jazzy singh <jazzy_gill@yahoo.com> wrote:<br>>> I was running a successfull asterisk box but i don't know what happened<br>>> for some reason asterisk shuts down as soon as I start it. It won't give me<br>>> any error or anything, but last nite when I was forking through error log I<br>>> did see this though<br>>> Jul 25 23:02:55 NOTICE[8329] chan_zap.c: Got event 18 (Ring Begin)...<br>>> Jul 25 23:02:55 WARNING[6651] channel.c: Avoided initial deadlock for<br>>> '0x8a94f88', 10 retries!<br>>> Jul 25 23:17:39
NOTICE[9303] chan_zap.c: Got event 18 (Ring Begin)...<br>>><br>>> and then i tried to recompile asterisk and it gave me this<br>>><br>>> codec_zap.c: In function ?find_transcoders?:<br>>> codec_zap.c:833: error: variable ?info? has initializer but incomplete<br>>> type<br>>> codec_zap.c:833: warning: excess elements in struct initializer<br>>> codec_zap.c:833: warning: (near initialization for ?info?)<br>>> codec_zap.c:833: error: storage size of ?info? isn?t known<br>>> codec_zap.c:838: error: ?ZT_TCOP_GETINFO? undeclared (first use in this<br>>> function<br>>> )<br>>> codec_zap.c:848: error: ?ZT_TRANSCODE_OP? undeclared (first use in this<br>>> function<br>>> )<br>>> codec_zap.c:833: warning: unused variable ?info?<br>>> make[1]: *** [codec_zap.o] Error 1<br>>> make[1]: Leaving directory `/downloads/asterisk-1.2.20/codecs'<br>>> make: *** [subdirs] Error
1<br>>><br>>><br>>> I was able to compile and install the same source code before. Please<br>>> someone help me if I don't get this to work today I might get fired :(..<br>>> please help.....<br>>> thanks in advance<br>>><br>>><br>>> ------------------------------<br>> <br>> <br>> <br>> <br>> <br>> <br>> <br>> ____________________________________________________________________________________<br>> Got a little couch potato? <br>> Check out fun summer activities for kids.<br>> http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&cs=bz<br>> <br>> <br>> ------------------------------------------------------------------------<br>> <br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<br>> <br>> asterisk-dev mailing list<br>> To UNSUBSCRIBE or update
options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-dev<br>> <br><br><br><br><br>------------------------------<br><br>Message: 6<br>Date: Thu, 26 Jul 2007 18:55:45 -0400<br>From: Russell Bryant <russell@digium.com><br>Subject: Re: [asterisk-dev] asterisk-dev Digest, Vol 36, Issue 68<br>To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com><br>Message-ID: <46A92671.2000605@digium.com><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>jazzy singh wrote:<br>> I apologize if you thought that I'm wasting everyone's time <br>> overhere ...I'm not...I'm under a lot of frustration right now....I just <br>> want this thing to start working like it was till yesterday. I'm a <br>> newbie at asterisk so I'm discovering things as I go I figure out how to <br>> make it print error messages when i restart here'we go<br><br>I already asked you once. Please move to the asterisk-users mailing
list. <br>People could easily help you there. This list is for development discussion.<br><br>> version = I'm using asterisk-1.2.13 (it worked fine till last nite..when <br>> I was testing it and must've called 20, 30)<br>> my version of zaptel -1.2.10<br>> lumenvox7.5<br><br>I also already noted that you need to update to a version that isn't 9 months <br>old before reporting that there is a problem.<br><br>-- <br>Russell Bryant<br>Software Engineer<br>Digium, Inc.<br><br><br><br>------------------------------<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by http://www.api-digital.com--<br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-dev<br><br>End of asterisk-dev Digest, Vol 36, Issue
69<br>********************************************<br></asterisk-dev@lists.digium.com></russell@digium.com></jazzy_gill@yahoo.com></asterisk-dev@lists.digium.com></harish.kasiviswanathan@gmail.com></asterisk-dev@lists.digium.com></eric@fnords.org></jazzy_gill@yahoo.com></asterisk-dev@lists.digium.com></kpfleming@digium.com></asterisk-dev@lists.digium.com></dboyd@ignitetrx.com></asterisk-dev@lists.digium.com></dboyd@ignitetrx.com></email@mattruby.com></asterisk-dev@lists.digium.com></russell@digium.com></blockquote><br><p> 
<hr size=1>Be a better Globetrotter. <a href="http://us.rd.yahoo.com/evt=48254/*http://answers.yahoo.com/dir/_ylc=X3oDMTI5MGx2aThyBF9TAzIxMTU1MDAzNTIEX3MDMzk2NTQ1MTAzBHNlYwNCQUJwaWxsYXJfTklfMzYwBHNsawNQcm9kdWN0X3F1ZXN0aW9uX3BhZ2U-?link=list&sid=396545469">Get better travel answers </a>from someone who knows.<br>Yahoo! Answers - Check it out.