similar to: question of flite installation

Hello I found following issue while trying to load flite module from CLI module load app_flite.so Unable to load module app_flite.so Command 'module load app_flite.so ' failed. [Jun 19 11:31:12] WARNING[23507]: loader.c:384 load_dynamic_module: Module 'app_flite.so' did not register itself during load [Jun 19 11:31:12] WARNING[23507]: loader.c:662 load_resource: Module

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is probably a newbie question: I'm trying to get the asterisk weather app at http://nerdvittles.com/index.php?p=136 installed on Centos 4.2 x86_64 SMP ( Linux localhost.localdomain 2.6.9-42.0.3.ELsmp #1 SMP Fri Oct 6 06:28:26 CDT 2006 x86_64 x86_64 x86_64 GNU/Linux ) rpm -ihv flite-1.3-1.aah.i386.rpm seems to complete successfully rpm -ihv app_flite-0.3-1.aah.i386.rpm seems to

HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

Version 2.1 of app-espeak and app-flite modules just got released. Changes include voice selection for flite, support for 16kHz file playback in asterisk 1.6.x, use of libsamplerate for faster resampling, better error handling and cleaner code. You can get them here: Asterisk espeak module: http://zaf.github.com/Asterisk-eSpeak/ Asterisk flite module: http://zaf.github.com/Asterisk-Flite/

Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

Hello, I recently purchased the Cepstral 6 text-to-speech engine (swift), and am now wondering if I should have bought something else. I would like to use Cepstral text to speech like some people use the Festival() or Flite() applications. For example, when I do a "core show application flite" at the CLI, flite is available to me: localhost*CLI> core show application flite

Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango

Hi, I wonder how I can relate the CDR records for the case of call transfer. I can't find their relationship in CDR. Any can advice? ango

Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango

Hi, I have an intel mother board dg43b (http://support.intel.com/Products/Desktop/Motherboards/DG43NB/DG43NB-overview.htm) which have an on board lan interface. In default, it can't be activated after CentOS5.2 installed. I can't find any driver or information about how to activate this interface. Can anyone tell me how to work it out? Thanks, ango

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still

Thanks. In app_realtime, it is very easy to get a value of a field by only applying the realtime application. However, in func_realtime, we need to get the key-value pair according to the position of it by using function CUT. After that, we need to apply another CUT to get the value. It will cause the following problems. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime 1.

Hi all, I found that a new field "lastms" is used in 1.4.24. What is the usage of that field and the datatype of it? ango

Hi all, I have read many forums and discussion groups talking about fax support in asterisk. Some of them conclude that asterisk doesn't support fax. However, some of them conclude that there is no relationship between fax and asterisk as asterisk will only pass the fax signal to the fax machine. I have tried the fax in asterisk before but failed. Anyone can give me some guideline how to

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-bounces at