similar to: AMI and extensions.conf

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

Hi people, I need to update the voicemail.conf from the UpdateConfig Action (AMI). The problem is that I executed: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-000000:append Cat-000000:test Var-000000:exten Value-000000:>999,test But I don't see the changes in the file. Can anybody tell me if there is something wrong in that code? Thanks,

Hi, I'm executing some commands using AMI... I suppose the log is saved in some place, but I don't know where... where is it saved?More details: I'm executing a UpdateConfig in the voicemail.conf file, but the file is not updated, so I would like to know why...Thanks, Anahi Anahi Ludue?a _________________________________________________________________ Ahora

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

Hello, I have a queue with only one element and one agent member. I want that my call leave the queue after 30s. My problem is that my call stays 60s in the queue and my agent is called 2 times. Can you say me how can i do it please?? -------------------------------- [queue] music=default strategy=ringall timeout=30 maxlen=1 context=mbdsys announce-frequency=0 announce-holdtime=no

Hello list, I'm using a PHP script to communicate with asterisk via AMI, and edit configuration files. So far everything went ok, but I came up with a little problem editing extensions.conf using 'updateconfig'. Is it possible to edit an existing line in extensions.conf file?, e.g. Given a 'category' in my extensions.conf as follow: [macro-mytest] exten =>

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

> From owner-icecast@xiph.org Wed May 5 16:50:59 2004 > Are you complaining that your station is too popular, or that ghost > clients aren't being disconnected for some reason, but no one is > actually listening? Sure, there are clients actually listeing too, but I do not beleive that some are listening for hundreds of hours :) What I would like is a 6 to 12 hours max connection

Hi, Does Asterisk manager allow multiple clients to connect to an Asterisk instance using the same user account ? Thanks,

Hi, I installed CentOS 3.3 and kept the machine up2date. I installed poptop and the newest ppp-cvs version ( ppp-2.4.3-0.cvs_20040527.6) We are running a ssh + ppp tunnel for our internet tunnel/tunnel to our NOC. After the upgrade we see these error messages while starting the tunnel: Jan 24 15:00:40 natgw modprobe: modprobe: Can't locate module ppp-compress-26 Jan 24 15:00:40 natgw

I'm not looking to move virtual machines between physical platforms; just want some protection between applications sharing a physical server. They'll all be using local disk in one form or another. Much the easiest to manage is just having the virtual machine use a file in dom0 as its disk. Without pre-allocating, this lets me over-commit somewhat to cover unknown future needs, for

Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

Hello, With this program I try to repeat analysis for different years. The results of the analysis are not printed when in the loop, except for the year sequence. What is wrong? Thanks a lot. for (i in 92:99){ cat("\n", "=============================================================\n", "YEAR =",i,"\n",

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour

Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : anthony at mbdsys.com Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46

I've mentioned this problem before but put off doing anything about it and maybe now someone can suggest the best solution. I have a 3-member RAID1 set where one of the members is periodically swapped and rotated offsite. The filesystem contains a backuppc archive which has millions of hardlinks that make it impractical to copy with a file-oriented approach. The current filesystem is

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not

I threw together a web gui for the static database configuration over the last couple of days. I built it using mod perl and the template toolkit. If enough people show an interest in this I'll put up a distribution, although it could take a few days. The interface is as generic as possible so you can throw pretty much any asterisk .conf file in and it works. The interface assumes you