similar to: Problem developing my office

Displaying 20 results from an estimated 7000 matches similar to: "Problem developing my office"

2005 Oct 11
1
noise when passing trougth speex_preprocess
Hi all, as in subject, speex_preprocess inject noise in my data. Someone can help ? Here's the way that i'm using: #define NN 160 /* 20msec di audio */ #define AUDIO_SAMPLERATE 8000 spx_int16_t TEMP_Buffer[NN]; speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); c = denoise; speex_preprocess_ctl(speex_pp_state, SPEEX_PREPROCESS_SET_DENOISE,&c); c = agc;
2005 Jul 22
1
Problem with Zaptel FXO..
Hi all, i've installed AMP and Asterisk following the INSTALL file and i have a problem with the TDM04B with 4 FXO: [root@srvoip ~]# ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default)
2005 Sep 22
1
Noise :-(
Hi all, i use speex preprocessor features in this way: =================================== #define NN 160 /* 20msec di audio */ ... int tbc=0,c,d,ret; spx_int16_t TEMP_Buffer[NN]; char DLECODE; /* Inizializza il preprocessore Speex se non inizializzato */ if(Modem->speex_pp_state == NULL) { Modem->speex_pp_state = speex_preprocess_state_init(NN,AUDIO_SAMPLERATE); }
2005 Sep 06
2
Going crazy with FAX :-(
I've upgraded Asterisk from CVS, spandsp and app_txfax and app_rxfax but i'm still unable to send/receive faxes :-(. I'm using amp_fax to send and this is what i get from logs: Sep 6 11:02:52 VERBOSE[10750]: -- Attempting call on Zap/g1/666 for application txfax(/var/tmp/ast_fax-1125997371.10240.1804289383.0|caller) (Retry 1) Sep 6 11:02:52 DEBUG[10750]: Dialing
2005 Jun 22
1
Newbie - Encoding PCM
Hi all, i've to encode voice from a voicemodem. I choose speex 1.0.5 for its quality in voice encoding. I've tried to implement an encoder but unsuccesfully. Here's my code: /* ============ SPEEX stream ENCODER ============================================ */ int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { /* buffer point to the
2016 Nov 30
2
Asterisk 14.2 CLI don't show debug/verbose data
Hi all, after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show what's happens. I've trying setting debug and verbose to 100 but nothing, no show. All commands works as expected but i can't what's happens on my asterisk server. asterisk*CLI> core show settings PBX Core settings ----------------- Version: 14.2.0 Build Options:
2005 May 17
1
sip show registry empty ?!?!!?
Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No
2015 Sep 16
4
Realtime Voicemail MWI
Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without
2005 Jun 30
0
speex_encode segfault
Hi, i'm following encoder example in the manual.pdf of speex documentation. Here's my portion of code: int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { int ret,c,d=0,nbBytes,ttBytes=0; float PCM_F[160]; char cBits[200]; #ifndef DISABLESPEEX speex_bits_reset(&IDA->speex_bits); for(c=0;c<num_samples;c++) {
2005 Aug 09
1
Incoming call action based on trunk
Hi all, i've asterisk with 8 FXS module connected to 8 PSTN lines. Each line have it's own number anche i want to do different action based on incoming call. For example, if call is on Line 1 i want to redirect it to extension 203, on line 2 to extension 201 etc etc it's possible ? How ? Looking in AMP i'm using AMP to manage Asterisk) this is not possible... Thanks ! Oz --
2005 Aug 29
0
Asterisk truncate my FAX !!!
Hi all, i've a problem receiving faxes. I'm using AMP and i hope that all work well without big changes. However i've done some tests on .tif file created by asterisk and i've noticed that it truncates my fax almost after 5-6 seconds. As results my pdf are corrupted and i receive a mail with empty pdf :-( someone can help me ? Thanks !!! Oz -- ---- O-Zone ! No (C) 2005 WEB
2005 Sep 08
0
sending fax....i'm in trouble !
hi all, i've this problem with app_txfax. Here's the log of the error: Sep 8 13:28:55 VERBOSE[10750]: -- Attempting call on Zap/g1/2430 for application txfax(/var/tmp/ast_fax-1126178934.10240.1804289383.0|caller) (Retry 1) Sep 8 13:28:55 DEBUG[10750]: Using channel 3 Sep 8 13:28:55 WARNING[10750]: Unable to allocate channel structure Sep 8 13:28:55 NOTICE[10750]: Unable to
2005 Jul 03
0
speex_encode segfault
Hi, i'm following encoder example in the manual.pdf of speex documentation. Here's my portion of code: int SPEEX_EncodePCM(struct _IDA_ClientSocket *IDA,char *buffer,unsigned char *PCM,int num_samples) { int ret,c,d=0,nbBytes,ttBytes=0; float PCM_F[160]; char cBits[200]; #ifndef DISABLESPEEX speex_bits_reset(&IDA->speex_bits); for(c=0;c<num_samples;c++) {
2005 Aug 10
2
Calling Extension directly
Hi all, i'm using Asterisk with several extensions with 7 PSTN lines. Is possible, for a caller, to dial directly an extensions ? For example, dial something like [PSTN number]*[ext number] ? Thanks ! -- ---- O-Zone ! No (C) 2005 www.zerozone.it
2005 Sep 15
0
TxFAX don't work
As subject, (i've updated spandsp to latest version) and this is the log: Sep 15 13:06:50 VERBOSE[14085]: -- Attempting call on Zap/g0/2479 for application txfax(/var/tmp/ast_fax-1126782409.10240.1804289383.0|caller) (Retry 1) Sep 15 13:06:50 DEBUG[14085]: Using channel 1 Sep 15 13:06:50 DEBUG[14085]: Dialing '2479' Sep 15 13:06:50 DEBUG[14085]: Deferring dialing... Sep 15
2015 Mar 04
3
supermin on arm
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I was testing oz/imagefactory on 32 bit arm, you have to have kernel-lpae installed to run kvm. while you can have the regular kernel installed also. You end up having the system booting the regular kernel and you do not get kvm. Ideally supermin will work with the lpae kernel.
1995 Dec 28
0
No subject
>From owner-majordomo Wed Dec 27 18:00:16 1995 Received: from minnie.cs.adfa.oz.au (minnie.cs.adfa.oz.au [131.236.21.160]) by freefall.freebsd.org (8.7.3/8.7.3) with SMTP id SAA02351 for <freebsd-announce@freefall.FreeBSD.org>; Wed, 27 Dec 1995 18:00:13 -0800 (PST) Received: (warren@localhost) by minnie.cs.adfa.oz.au (8.6.8/8.3) id MAA04468; Thu, 28 Dec 1995
2011 Aug 22
0
ANNOUNCE: oz0.6.0 release
All, I'm pleased to announce release 0.6.0 of Oz. Oz is a program for doing automated installation of guest operating systems with limited input from the user. Release 0.6.0 is a bugfix and feature release for Oz. Some of the highlights between Oz 0.5.0 and 0.6.0 are: - The ability to specify the destination for the ICICLE output from oz-install and oz-generate-icicle - pydoc
2013 Jul 29
0
Re: ANNOUNCE: Oz 0.11.0 release
On 07/28/2013 09:43 PM, Chris Lalancette wrote: > All, > I'm pleased to announce release 0.11.0 of Oz. Oz is a program > for doing automated installation of guest operating systems with > limited input from the user. Release 0.11.0 is a bugfix and feature > release for Oz. Some of the highlights between Oz 0.10.0 and 0.11.0 > are: > > * Add support for
2014 Jan 03
0
ANNOUNCE: Oz 0.12.0 release
All, I'm pleased to announce release 0.12.0 of Oz. Oz is a program for doing automated installation of guest operating systems with limited input from the user. Release 0.12.0 is a bugfix and feature release for Oz. Some of the highlights between Oz 0.11.0 and 0.12.0 are: * Fixes to concurrent oz-install invocations * Python 3 compatibility in the test suites * Support for Ubuntu