similar to: sending fax....i'm in trouble !

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

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

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 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

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

I'm running Postfix as my email server. When comes to configuring Asterisk Fax, Inter7 recommends QMail. Does anybody knows postfix equivalent for the qmail? I don't have any knowlegde in QMail. This is the installation guide for astfax. ( http://www.inter7.com/astfax/INSTALL ) ------------------------------------------------------------------------------ 3) Configure MTA [ This step is

Has anyone configured ast_fax (sending faxes via asterisk) with sendmail? The creation of rules to trap all numbers <mailto:XXXXXXXXXXX@domain.com> XXXXXXXXXXX@domain.com seems too complicated. Does anyone have setup details to share? (I don't want to switch MTA's). As a workaround, I could launch the app automatically from sendmail using an alias like: fax: "|

On Friday 07 October 2005 17:48, Michael Stahl wrote: > Roman: > > I created two bash scripts called Mail2Fax and Fax2Mail for use with the > asterisk sever. > > They leverage the app_txfax and app_rxfax scripts, along with ast_fax. > They make using these apps a lot easier, including being able to mail to > fax@domain.ca for outgoing faxes and then extracting phone numbers

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 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, We are trying to use astfax (from Inter7) to send faxes directly from email accounts through *. The problem is that the fax machine answer the call when we test astfax (with the line in the README) and then it retries three or four times to send it, and hangup the channel. The fax is connected to a zap channel, and it has an extension configured for itself. This is an excerpt of the

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

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++) {

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, 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