similar to: Record Application Problem

Followed this; #cd /usr/src #export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot #cvs login (password is anoncvs) #cvs checkout zaptel libpri asterisk #cd zaptel ; zaptel equipment #make clean; make install #cd ../libpri ; isdn #make clean; make install #cd ../asterisk #make clean ..but

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

Does someone have the settings for 'indications.conf' in Portugal? Thank you, Pedro Goncalves ---------------------------------------------------------------------------- -- Pedro Goncalves PT Inova??o SA - P?lo do Porto Largo de Mompilher, 22 - 4? 4050-392 Porto - Portugal Phone: +351 222079329 Email: est-p-bgoncalves@ptinovacao.pt

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

Dear All, This is my first post to this mailing list (and yes, I did read http://www.r-project.org/posting-guide.html ) so please forgive any faux pas. I'm trying to make a visualization that looks like this http://www.gradient-da.com/img/temperature%20surface%20plot%20470x406.JPG(found through google). The idea is to have a 3D surface plot overlapping a 2d representation of a surface. I can

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

Thanks Jim, But i didnt find this project... It's in CCS folder or on TI site ????? Thk's Em 08/04/2008, ?s 15:52, Jim Crichton escreveu: > The TI directory of the Speex source distribution contains a C5509A > project that builds and runs in TI's Code Composer Studio > simulator. This project does file I/O to files specified in the > main source file. See

Hello! I sent this message to samba-technical yesterday but as I saw no response (not even a RTFM ;), I'm crossposting. I just need some help to track down this bug. I can write a patch if I can find the guilt... ----- Forwarded message from "Luis Claudio R. Goncalves" <lclaudio@conectiva.com.br> ----- Date: Thu, 29 Nov 2001 12:34:41 -0200 From: "Luis Claudio R.

and a little more on fax config support ----- Original Message ----- From: "Tilghman Lesher" <tilghman@mail.jeffandtilghman.com> To: <asterisk-users@lists.digium.com> Sent: Monday, August 11, 2003 4:48 PM Subject: Re: [Asterisk-Users] Fax Handled > On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote: > > On Mon, 11 Aug 2003 15:15:08 -0500 > > >

The TI directory of the Speex source distribution contains a C5509A project that builds and runs in TI's Code Composer Studio simulator. This project does file I/O to files specified in the main source file. See README.TI_DSP in the main directory for some tips. You should use the 1.2 beta 3 distribution. You should be able to load, build, and run this with no effort except to get the

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

A SipPulse acaba de liberar o TFPS (www.tfps.co), solu??o para combate a fraudes de fomento de tr?fego internacional em telefonia. O sistema ? capaz de detectar 99.99% das tentaivas de fraude em tempo real. Durante o Webinar, abordaremos como proteger servidores Asterisk e Elastix/FreePBX de fraudes, medidas basicas como configura??o de firewall e remo??o de servi?os desnecess?rios e em seguida

Hi, I'm using Dovecot with LDAP. My users have an home directory and a field named 'mailMessageStore'. Then the full mailbox path is 'homeDirectory/mailMessageStore' In dovecot-ldap.conf i configured as follows : user_filter = (&(objectClass=qmailUser)(cn=%u)) user_attrs = qmailUID=uid,qmailGID=gid,homeDirectory=home,mailMessageStore=mail=maildir:~/%$ Everything is

Hi, I'm use a DSP C5510 to implement a solution with voice, but I'm a beginner in the world of DSP, any one has a example or documents about how to use a speex in DSP's ???? Thanks. Att.; Rafael Vieira Gon?alves skype: rafaelvieira.goncalves msn: tux_surf at hotmail.com email: daconfama at gmail.com P Antes de imprimir, pense em sua responsabilidade e compromisso com o

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card