similar to: Set(LANGUAGE()=language throwing warnings

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

i have a problem with AGI in Asterisk 1.0.5, the problem occurs either with PHP or C AGI scripts/programs. Well, its simple, either asterisk is not sending correctly the command responses to the standard output, or for some unknown reason to me the scripts/programs are not able to read it from standard input. I have the next C test program for AGI: #include <stdio.h> main() { char

I'm writing an AGI script and want it to dial a number on a channel connected to the PSTN. It would look something like this (pseudo-code follows): if ($a){ dial("8005551212"); }else{ dial("8665550000"); } The part I can't seem to get right is the "dial" function. I tried to mimic the dial plan like so sub dial($number){ print

Hi, I am using Asterisk: 1.6.1.20 LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (A104d) I handle some calls with my own PHP-AGI-Script. After a dial-command I use "GET FULL VARIABLE ${answeredtime}" or "GET FULL VARIABLE ${dialstatus}" and get valid information. Sometimes "dialstatus" has the value

This is release 1.0-rc1 of Wine, a free implementation of Windows on Unix. This is the first release candidate for Wine 1.0. Please give it a good testing to help us make 1.0 as good as possible. In particular please help us look for apps that used to work, but don't now. See http://wiki.winehq.org/PlatinumRegressionHunt for details. What's new in this release (see below for details):

Dear R users, I'm happy to announce the release of version 0.3 of the udpipe R package on CRAN (https://CRAN.R-project.org/package=udpipe). The udpipe R package is a Natural Language Processing toolkit that provides language-agnostic 'tokenization', 'parts of speech tagging', 'lemmatization', 'morphological feature tagging' and 'dependency parsing' of

Dear R users, I'm happy to announce the release of version 0.3 of the udpipe R package on CRAN (https://CRAN.R-project.org/package=udpipe). The udpipe R package is a Natural Language Processing toolkit that provides language-agnostic 'tokenization', 'parts of speech tagging', 'lemmatization', 'morphological feature tagging' and 'dependency parsing' of

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

Hola, Lo primero muchas gracias a todos por vuesta habitual ayuda. Llevo meses con las funciones lmer, pero tengo un problema no se como plantear unas comparaciones multiples una vez demostrado que el factor tiene diferencias significativas. Tengo multiples opciones, Solución A: una es hacer un bonferroni a saco (pero eso me chirria ya que tego muchos datos) Solución B: Otra es utilizar glht,

Hi there, admittedly this is a slightly aged RH system. ;-> Yet what I'd like to know is if this compile problem is clearly due to the gcc version, or if there is a way to solve the issue...? Cheers, Philipp [root@box src]# gcc -v Reading specs from /usr/lib/gcc-lib/i386-redhat-linux/2.96/specs gcc version 2.96 20000731 (Red Hat Linux 7.2 2.96-112.7.2) [root@box src]# rpm -q unixODBC

The Wine development release 1.2-rc3 is now available. What's new in this release (see below for details): - Many translation updates. - A lot of bug fixes. The source is available from the following locations: http://ibiblio.org/pub/linux/system/emulators/wine/wine-1.2-rc3.tar.bz2 http://prdownloads.sourceforge.net/wine/wine-1.2-rc3.tar.bz2 Binary packages for various

Hello all, I have a strange problem that I'd like advice on. Here I run several samba servers as home directory and print servers for an NT 4.0 (sp6a) domain. The domains relevant to this problem are CANNING and STUDENT. The STUDENT domain trusts users in the CANNING domain. My samba server is RedHat 6.2 (+ updates) running samba 2.2.7 (built today). I upgraded because of a different

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

On 4/20/2023 7:16 AM, Rhys Hanrahan wrote: > > Hi All, > > I’m having issues figuring out how to set no DTMF timeout on the AGI > GET DATA command as “0” still has a multi-second timeout if no input > is given. I am trying to find the source code of the AGI command to > figure it out, but I can’t for the life of me find the underlying > source code. The closest I’ve

Hi, Can any one help I'm trying to install asterisk on suse 9.3 pro from cvs release v1_0 version 1.0.9 and when I try to make from the asterisk directory I get the following error. Is there anybody that could give me a pointer as to what the issue may be? DDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN

Head of Department of Applied Statistics and Epidemiology - Institute of Cancer Epidemiology, Copenhagen, Denmark The Institute Population-based cancer research in the Nordic countries is in the international forefront. Since the foundation of the Danish Cancer Registry in 1942 the Institute of Cancer Epidemiology has played a significant role in this research area. The Institute is looking