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...d VarLib directory: /var/lib/asterisk Data directory: /var/lib/asterisk ASTDB: /var/lib/asterisk/astdb IAX2 Keys directory: /var/lib/asterisk/keys AGI Scripts directory: /var/lib/asterisk/agi-bin Any hint ? Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Universit? degli Studi di Siena tel: 0577.(23)5000 - centralino at unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it -------------- next part ---...

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

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 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

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 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, 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

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, 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 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 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

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

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

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202