[asterisk-users] DTMF CallerID

2010-11-24 Thread Antonio Modesto
Hi, Does anyone know if CID is already working with Digium TDM800P card using DTMF signalling? (I'm brazillian) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] Problems with ZAP Channels

2011-01-12 Thread Antonio Modesto
Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the soft hangup Zap/zapchannel in the asterisk CLI to release the channel. Here is my zapata.conf:

[asterisk-users] Digium WCTDM24XXP DTMF CallerID

2011-04-27 Thread Antonio Modesto
Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -- _ --

[asterisk-users] How to continue processing a context after a Hangup

2011-06-02 Thread Antonio Modesto
Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the

[asterisk-users] Asterisk Online Training

2011-06-06 Thread Antonio Modesto
Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Digium WCTDM24XXP DTMF CallerID

2011-06-07 Thread Antonio Modesto
console: Incoming call from via DAHDI/1-1 i've put this line in my incoming context to show the message above: Verbose(Incoming call from ${CALLERID(num)} via ${CHANNEL}); Thanks On Wed, 2011-04-27 at 09:58 -0500, Shaun Ruffell wrote: On Wed, Apr 27, 2011 at 09:26:24AM -0300, Antonio Modesto

[asterisk-users] Problems with DTMF Caller ID

2011-07-08 Thread Antonio Modesto
Good afternoon, I'm trying to configure my asterisk to work with DTMF signaling (I live in Brazil) , i've put these lines in my chan_dahdi.conf usecallerid=yes callerid=asreceived cidsignaling=dtmf cidstart=polarity my dahdi system.conf loadzone=br

[asterisk-users] Macro to Dial a Channel Group using Round-robin

2011-07-20 Thread Antonio Modesto
Good morning, I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but i'm having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, until the call be answered. For example, i have

Re: [asterisk-users] Macro to Dial a Channel Group using Round-robin

2011-07-20 Thread Antonio Modesto
Oh man, so easy, thank you very much! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

[asterisk-users] Problems with System() application

2011-07-20 Thread Antonio Modesto
Good afternoon, I am trying to use the System() application but it is always returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run this command: System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt} ${exten}); This is the content of the

[asterisk-users] [SOLVED] Re: Problems with System() application

2011-07-20 Thread Antonio Modesto
The problem was the directory which i was writing the logs, i put the log file in /var/log/asterisk and it worked. Thanks. On Wed, 2011-07-20 at 13:03 -0500, Jorge GutiƩrrez wrote: Are you able to execute: log.sh through the asterisk user? On Wed, 20 Jul 2011 14:53:53 -0300, Antonio

[asterisk-users] Callback + DISA

2011-07-26 Thread Antonio Modesto
Hello, I am trying to use a Callback system that return the call to some number then give it a dial tone with DISA. The callback works well and i can hear the dial tone, the problem is that DISA doesn't do anything when i press any extension number of the current context and hangs the call up

Re: [asterisk-users] Why no traction for Windows version?

2011-07-27 Thread Antonio Modesto
On Tue, 2011-07-26 at 09:45 +0200, Gilles wrote: On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon) soeren.malc...@mcon.net wrote: And asterisk just runs fine on linux why bother ? Because I, for one, would like to run Asterisk on my Windows workstation at home as an enhanced

[asterisk-users] Problem with Atxfer for the calling party

2011-10-31 Thread Antonio Modesto
Good Morning, I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it is working well so far, i'm just having some problems with atxfer. I have written this macro to dial sip extensions: macro dial_sip(exten) { Verbose(2,== Chamando a MACRO dial_sip - ponto 1

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-11-01 Thread Antonio Modesto
this context doesn't have the ramais context included. Is there some way to specify on which context the macro will run? On Mon, 2011-10-31 at 09:09 -0200, Antonio Modesto wrote: Good Morning, I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it is working well so far, i'm

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-11-21 Thread Antonio Modesto
Hi There, I'm still having this problem, Does somebody know what can be happening? Regards. On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote: Hello, The exten is the parameter passed to the macro, which contains the sip device name. I'll change the name to another less

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-12-12 Thread Antonio Modesto
Nothing? On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote: Hi There, I'm still having this problem, Does somebody know what can be happening? Regards. On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto wrote: Hello, The exten is the parameter passed

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-12-13 Thread Antonio Modesto
rewrite all my macros in the common way, it will work, but that's a lot of coding for me. On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote: Nothing? On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote: Hi There, I'm still having this problem, Does somebody

[asterisk-users] AEL x LUA

2011-12-13 Thread Antonio Modesto
Hello, I would like to receive some suggestions about dialplans written in lua, actually my dialplan is written in ael, but i'm having some problems with it. I noticed that asterisk just translates ael to the old extension language, does it do the same with lua? Thanks --

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-12-13 Thread Antonio Modesto
: Set(__TRANSFER_CONTEXT=${MACRO_CONTEXT}); Is it right? Thanks. 2011/12/13 Antonio Modesto mode...@isimples.com.br Hello everybody, I found that if i write my macro in the extensions.conf (not in ael), the atxfer works well

Re: [asterisk-users] Problem with Atxfer for the calling party

2011-12-13 Thread Antonio Modesto
the value to my extensions context and it worked fine. Thanks. 2011/12/13 Antonio Modesto mode...@isimples.com.br Hello everybody, I found that if i write my macro in the extensions.conf (not in ael), the atxfer works well, the problem is that ael

[asterisk-users] Problems with DAHDI Channels

2011-12-16 Thread Antonio Modesto
Hi everybody, I'm having a problem with some of my DAHDI Trunks, it is a strange thing, above is the output of the core show channels command: Channel Location State Application(Data) DAHDI/11-1 ~~s~~@dial_dahdi:15 Up Dial(DAHDI/12/

Re: [asterisk-users] Problem with Atxfer for the calling party [SOLVED]

2011-12-20 Thread Antonio Modesto
As explained in the posts before, this tread was solved. Thanks. On Tue, 2011-12-13 at 17:07 -0200, Antonio Modesto wrote: On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote: Hi Antonio, I'd never had used extensions.ael but in extensions.conf, using Macro I always set

[asterisk-users] Stuck DAHDI Channels

2012-01-13 Thread Antonio Modesto
Good afternoon, My current Asterisk is a FreeBSD 8.2-STABLE machine with asterisk18-1.8.7.1 installed from ports collection and . My old asterisk was a pre-configured system (Brazilian Disc-OS), and on both systems i have the same problem, sometimes some of my DAHDI channels get stuck, here

[asterisk-users] Stuck DAHDI Lines

2012-02-09 Thread Antonio Modesto
Hi, Sometimes some of my dahdi channels become stuck, It is very strange, here is the output of the core show channels command: pabx*CLI core show channels Channel Location State Application(Data) Local/104@ramais-cc0 104@ramais:1 Up

[asterisk-users] Asterisk trying to call a queue with no members

2012-07-06 Thread Antonio Modesto
Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example, if I create queue called test, which has no members logged in, and try to place a call using Queue(test) I get into

Re: [asterisk-users] Asterisk trying to call a queue with no members

2012-07-06 Thread Antonio Modesto
On Fri, 2012-07-06 at 11:09 -0500, Kevin P. Fleming wrote: On 07/06/2012 10:15 AM, Antonio Modesto wrote: Hi, I am trying to configure some static queues in asterisk, it's almost working, the problem is that asterisk is not verifying if the queue has logged members. For example

Re: [asterisk-users] Asterisk trying to call a queue with no members

2012-07-06 Thread Antonio Modesto
On Fri, 2012-07-06 at 13:32 -0500, Kevin P. Fleming wrote: On 07/06/2012 12:36 PM, Antonio Modesto wrote: I don't want the users to manually login in the queue, I want they join the queue when they turn on their phone. I thought that this was the right way of doing it, how can I do

[asterisk-users] Queue timeoutpriority=app doesn't working as explained in conf.sample

2012-07-09 Thread Antonio Modesto
Hi, I am trying to understand how the asterisk queues timeout works. I want to set a timeout for each phone ring, and a maximum timeout a caller can wait in a queue, but I didn't get it working the way I want. My Queue application call looks like this: Queue(myqueue,t,,,60); and my

[asterisk-users] Changing auto mixmonitor output file name

2012-07-16 Thread Antonio Modesto
Hi, I want to enable call recordings by simply pressing the recording key defined in features.conf. The problem is that I didn't find a way to change the name of the output file, the Dial application has two options for enabling call records, the w and W and x and X. What is the diference

[asterisk-users] DAHDI and Tiger320 Chip

2012-10-25 Thread Antonio Modesto
Hi, I've got a ISDN Interface: *Tiger Jet* Network Inc. Tiger3XX Modem/ISDN interface, I'm trying to use it with DAHDI 2.6 but it doesn't work, I'm thinking that dahdi doesn't support this device, I've loaded all of available dahdi drivers and none of them worked. Does anybody know what I can do

[asterisk-users] Asterisk dropping calls on transfer

2013-07-25 Thread Antonio Modesto
Hi, I'm having a weird problem with asterisk (asterisk18-1.8.12.2_1). Every call on the system, whatever it comes from the PSTN or from local extensions, when we hit the '#' button to transfer the call, asterisk just disconnects it, without any error or log, here is my current features