similar to: Realtime Voicemail MWI

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:

Hello Everyone. In my outbound contexts, I'm using "${DIALSTATUS}" to fail over to other routes if the chosen route rejects the call. Now, My current scenario is if I get "BUSY" back from the first provider, I send a busy back to my customer. If I get something like CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt the call. This works

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

Hello, is it possible to disable the CDR record creation for all dialed parties? From my limited testing it looks like CDR_PROP(disable) is effective only for the first party (the one specified before the first ampersand in the Dial application argument) and I can't find any way to disable it for the other ones (I think the CDR in question is written after the Dial completes). Is it by design?

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

Hey everyone, Currently, I've got a PBX that is emailing me on call failures to an international SIP provider of ours. I'm doing this with exten => 1,1,System(mail -s "Call from ${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}" nick at flhsi.com < /dev/null) This works fine, However it's a little lacking. For Instance, Our INTL SIP

Hello All, Looking for a little guidance on Real Time Pattern Matching. We are attempting to block outbound 411 via when someone dials NXX-555-XXXX, The must common being NXX-555-1212. However, We have some outbound providers that consider any call to NXX-555-XXXX a directory assistance call. So simply making my pattern _NXX5551212 doesn't work. So as you can see from the lines

I''ve installed a bering box acting as a firewall for a lan; the lan is 192.168.1.0/24 the bering box is 192.168.1.254 I''ve installed a squid server 192.168.1.1 It is possible to configure shorewall for a transparent proxy to the squid server? I''ve tryed with REDIRECT loc loc:192.168.1.1:3128 tcp www - !192.168.1.1 in the rules file I get this error: Error:

Alle, Does anyone know if there is/where to get an rpm of flow-tools V0.66 or better for CentOS/RHEL 5.3? We've been trying to build from the SRPM @ http://cng.ateneo.net/cng/wyu/software/srpm/flow-tools-0.68-2.src.rpm with no luck. Best Regards, Camron -- Camron W. Fox Hilo Office High Performance Computing Group Fujitsu Management Services of America, Inc. E-mail: cwfox at

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 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, 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 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)

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

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

Dear list members, First of all thank you for your helpful advices. After your answeres to my firt mail I studied a lot (R-News n?5) and I tried to perform my analysis: First, to fit a GLM with a nested design I decided to use the function "lmer" in package "lme4" as suggested by Spencer Graves and Filippo Piro. I remember to you that my data were: land use classes, 3 levels

Hello everybody, after upgrading dovecot on a debian wheezy installation from the standard package version (dovecot 2.1.7) to dovecot 2.2.13-11 from wheezy-backports, i noticed some errors in my logs... Apr 28 22:00:13 lmtp(4879, xxxxxxxxx at unipd.it): Info: copy from <lmtp DATA>: box=INBOX, uid=error, msgid=<20150428200011.47D801F32 at mydoom.unipd.it>, size=1523 They are not

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); }

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Just to let you know: I'm planning on releasing v2.1.11 today/tomorrow. If you wish to get something fixed for it, ask quickly. :)