similar to: Asterisk truncate my FAX !!!

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); }

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;

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

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

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

Hi all, i need some advices. In my office we have 7 PSTN lines from central phone-office (one line - one number) and we plain to install an Asterisk server as PBX. We need to have 15 PSTN devices (phones, fax, etc) in opur office. I've seen FXS and FXO but i'm not sure: we need 7 FXO and 15 FXS ?!?!?!?!? There's a smarter solution ? Thanks ! Oz -- ---- O-Zone ! No (C) 2005

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)

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++) {

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

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 --

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++) {

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

Hello. =) I''m having a bit of trouble with my nfs xen installation and have spent much time trying to research/figure out this problem with no luck, so I''m hoping someone might be able to help me here. =) I''ve successfully installed xen 3.02 on Linux RH4 on my Sun single proc opteron v20z server. I''ve installed what I believe to be a base Linux OS on a

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

Hi, i added a native (client) Socks V4/V5 support inside FreeBSD libc library. The work is based of my project (see http://csocks.altervista.org) CSOCKS. You can get it here: http://csocks.altervista.org/download/FreeBSD_libc.tar.gz CHANGES: I changed the file: /usr/src/lib/libc/Makefile I added the Directory: /usr/src/lib/libc/socks They contains the files:

Upgrade: 1) Added IPv6 Support (need to be tested) Cheers Raffaele Hi, i added a native (client) Socks V4/V5 support inside FreeBSD libc library. The work is based of my project (see http://csocks.altervista.org) CSOCKS. You can get it here: http://csocks.altervista.org/download/FreeBSD_libc.tar.gz CHANGES: I changed the file: /usr/src/lib/libc/Makefile I added the Directory:

Hi, R function aggregate can only take summary stats functions, can I aggregate text columns? For example, for the dataframe below, > a <- rbind(data.frame(id=1, name='Tom', hobby='fishing'),data.frame(id=1, name='Tom', hobby='reading'),data.frame(id=2, name='Mary', hobby='reading'),data.frame(id=3, name='John',

Here you go, Ian...... ---------- Forwarded message ---------- From: G3RIR <g3rir@yahoo.com> Date: 05-Apr-2006 20:54 Subject: [dmuars] Eh up - March 144 results altered To: dmuars@yahoogroups.com What's going on here. The results of the MArch 144 UKAC have been re-published and we have lost out considerably. Either I don't understand the rules or we have been robbed We scored

Hi, I just compiled and installed the latest compiz from git head and I have a strange issue with windows positioning: all newly created windows are placed at 0,0 (upper left corner) of the screen, so the title bar (and menu bar too) is under my gnome panel. I attach a screenshot to show the problem. I can't understand if this is a problem of compiz configuration (plugin order or something