similar to: CALLERPRES() with Queue

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

Hi List, When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22 and 1.4.35 respectively, I am getting multiple lines of this strange error: ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s): On three other servers with same versions of asterisk, i.e. 1.4.22, I don't see this error. Number of lines of the error are the same as the number of lines of the

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

I should have included the debug output: <PJSIP/Twilio-NA-W-3-In-00000006>AGI Rx << CALLERPRES(allowed) <PJSIP/Twilio-NA-W-3-In-00000006>AGI Tx >> 510 Invalid or unknown command -----Original Message----- From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of TTT Sent: Saturday, July 1, 2023 11:37 AM To: 'Asterisk Users Mailing List -

Hi All, I'm having some troubles with using call files. I'm trying to establish the following: - want to use call files to connect two (outside) extensions - want to use the outbound routes set in FreePBX - want to set the outgoing callerid for both calls - want to set a custom CDR field in MySQL ( field name 'azonosito' ) Asterisk is version 1.8.2.3 with freepbx 2.8.1. What

[Apologies in advance for the train of thought prose, but it is brainstorming after all…] I'm going to focus on self-descriptive names rather than literary or fantasy references… Advanced Compiler Kit, affectionately known as “ACK!”? It has an ill- deserved nod to NeXT, even. (Completely the wrong language, after all.) Core Compiler? Heh, I don't think that'd get past certain

Estimados, tengo que hacer un calculo muy simple, pero con una BD mas o menos grande (250mil filas x 500 columnas) ... esta BD esta en formato de SPSS y la importo asi: library(foreign) bdr <- read.spss("CASEN_2013_MN_B_Principal.sav", use.value.labels=FALSE, to.data.frame=TRUE) luego, quiero transformar el DF en un data.tale pues el calculo requiere obtener promedios de

The AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original

Hi, do anyone know which statistical software have the Brown-Forsythe Test for differences of numerical continue data groups wich variance heterogenity? not ofr evaluate the homogenity of variances, for evaluated the differences between groups as ANOVA. thanks Jos?? Herrera M??xico _________________________________________________________________ T1msn Search. Todo lo que buscas ahora m??s

Con data.table todo puede ir muy rapido. > require(data.table) > M=matrix(c(5,NA,NA,NA,6,NA,7,NA,8),3,3) > M [,1] [,2] [,3] [1,] 5 NA 7 [2,] NA 6 NA [3,] NA NA 8 > M2=data.table(M) > M2 V1 V2 V3 1: 5 NA 7 2: NA 6 NA 3: NA NA 8 > M3=melt(M2,variable.name = "columna") > M3 columna value 1: V1 5 2: V1 NA 3: V1

Hi i use this into my extension : exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)}) exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)}) exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)}) exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten =>

Hi List, it feels silly, but here I am. My Asterisk box is useless, after running a long delayed yum update (Centos box). ***** A few details on the box: cat /etc/redhat-release CentOS release 5.10 (Final) arch i686 uname -a Linux hermes 2.6.18-371.4.1.el5 #1 SMP Thu Jan 30 06:09:24 EST 2014 i686 athlon i386 GNU/Linux asterisk -r Asterisk 1.6.2.20, Copyright (C) 1999 - 2010 Digium,

I have been fighting with the ability to set the caller ID when I make outbound calls via a PRI line as well as via my SIP provider. The more I play around the less I understand. There is a setting in chan_dahdi.conf that seems to say do not pay attention to the CALLERPRES value and just allow the ID to be set. This setting is usecallingpres. If this is set to yes then the value of CALLERPRES

Hi We want optimize my extensions file conf on asterisk 11.4.0 : We have a big quantity of extensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten =>

Hola Javier, gracias por contestar tan rapido (tarde en la noche) ... olvide decir que, obviamente, he revisado la internet buscando y no encuentro ni siquiera la tabla de analisis de varianza del diseño ... habia revisado el paquete agricolae porque este trae diseños parecidos pero no exactamente lo que necesito. Sigo buscando, muchas gracias, Eric. On 29/04/15 22:57,

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

Hi, I've run into a minor snag trying to pass on CALLERID presence from one Asterisk to another via SIP (both running 13.16.0) I have a PRI coming in PBX_A and PBX_A is connected to PBX_B via SIP. PBX_A gets PRI calls on a 4 port Digium card, and each call naturally has its own callerid values and presence. I pass on those calls to PBX_B via SI, and I'm trying to pass on this

Hi All, In my dialplan and standard asterisk CLI logging i see that i am able to restrict the callerid when dialing out with asterisk. however, on the receiving phone, the callerid is still displayed. When i increment the logging of the pri with "pri set debug on span 1" on the CLI i also get the lower level debugging info from the pri.

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hola Jorge, muchas gracias por tu pronta respuesta, no me di cuenta que el formateo podría causar problemas, envío de nuevo el código sin formatos. La idea básica es para un set de números de columnas (desordenados) y un set de numeros de fila el loop lo que hace es ir a la fila y columna correspondiente de data, tomar el valor y luego hacer la media sobre esos.