similar to: playback soundfile stored in mysql database

Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated.Rgds,Akpome _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE!

thanks Billy. I replaced print "STREAM FILE $filename \"\"\n"; with print "EXEC PLAYBACK $filename \n"; and it worked fine. Interestingly when I did print "STREAM FILE beep \"\"\n"; within the script, it worked. If I wasnt a newbie to asterisk I wouldve thought this to be strange. >From: "William Piper"

I've been using it in a test environment with no problems. However, I haven't used it in production yet. I'm doing some voice broadcasting with a PRI and so far I'm content with the performance. -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Akpome Akpoguma Sent: Friday, February 03,

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

I downloaded recordPad and recorded a wav file and tried playback on asterisk got the same error as before -- WARNING [1225991360] Format.wav.c:132 check_header:unexpected header size 18-- when I recorded in gsm format on my laptop asterisk did playback well I used sox to resample the recorded wav file on the asterisk machine into wav again and asterisk playback worked well. The sound

Does anyone how you might detect a period of x milliseconds of silence using AGI ? Rgds, Stuart

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

Hello list, is it possible with the monitor-command to record conversations and place the soundfile in a pre-defined directory per user ?! So when extension 200 presses '*#' to record the conversation, the resulting sound file is written to his home directory on the Samba-server. This way each user has his own directory with its recordings that no one else can access (as default rights

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

hi all.....well ive installed each and everything according to the scratch installation but the problem is when i try to login a user through vicidialgui application it gives an error that there are 0 leads in the hopper to dial....well im pasting the result of a few queries .....plz if any one can do help me out .... this is what the error looks like... SELECT count(*) FROM vicidial_hopper

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

hello when i try to search for any word that contains the german characters it returrns an error. i have used the xapian database to store metadata of urls that are inturn scraped with the help of a crawler. how do i search for german characters without any error. Thanks _________________________________________________________________ Express yourself instantly with MSN Messenger! Download

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command "realtime mysql status" on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300