similar to: Cannot Dial Out From *

I am using the sample (slightly modified) macro for dialing phones. My extensions are in the 2000 range. Each extension has it's own external DID. Is there any way to have the macro look up the DID to be used for the extension and set the DID as the callerid? Maybe from a flat file somewhere? Or is there a better way to do this??? I know I can use callerid in sip.conf, but I only want the

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

In my sip.conf, under general I have: disallow=all allow=g729 allow=alaw allow=ulaw Then I have a specific sip: [RNK] <clip> disallow=all allow=alaw allow=ulaw allow=gsm If I do this: exten => _9.,1,Dial(${EXTEN}@RNK,60) The call still goes out as G729 even though I've told the RNK to disallow g729. I need to be able to make other 729 calls but to this one paticular group, they

When a user dials 999 he is always asked for the mailbox and has to enter his mailbox number and password. As I understand this shouldn't happen because the CALLERIDNUM is passed over to VoicemailMain. It's annoying to have to enter the number everytime ... The voice mail configuration is read from MySQL. We are using the CVS version from a few days ago. Extract from extensions.conf:

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

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

I have a server running SuSE 9.3 (Samba 3.0.13-1.1). The underlying filesystem is xfs, and the NICs are Netgear gigabit. 2 Gb of ram in a P4/3.0 Ghz. The workstations are windows XP Pro, with all service packs installed, on P4 3+ Ghz, 1-2 Gb of ram. (varies a bit by workstation) I have one particular tree on the server that contains over 12K files in a few hundred subdirs. Breaking it up

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

Dear R users, I want to share my joy with you. Please see the following excellent introduction to R "Statistiques avec R " by Vincent Zoonekynd http://zoonek2.free.fr/UNIX/48_R/all.html In paticular, you can see a lot of fascinating graphics examples of R from which you can get many hints. Soryy if this is already well-known, but the CRAN search did not show nothing with the keyword

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Hello friends of R list, Im a physician and Im not that good in statistics. I have posted similar email in the epi-sig list but no one aswered so far. Im cunducting a systematic review on two diagnostic test for a particular tropical disease. I found a free software that make most of the analysis called MetaDiSc. However there is a paticular analysis that I wuould like to try that it is not

Hi all! In my extensions I have the following: exten => 999,1,Answer() exten => 999,2,PlayBack(beeperr) In /var/lib/asterisk/sounds/ I have both beeperr.gsm & beeperr.ulaw, both with '-rw-r--r--' permissions. when I dial extension 999 I get: ************************************ -- Executing Answer("SIP/asterisk.domain.com-081477a0", "") in new stack

Hi, For some reason, I seem to have issues with dailing toll free numbers and can't seem to find out why, sometimes, I get a busy signal. Some other times I get weird errors from the phone. The error below was a simple busy signal. Here's couple of my info relevant to the problem: -- Reconfigured channel 1, PRI Signalling signalling -- Reconfigured channel 2, PRI Signalling

Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Hi Chandler, I have tested the new shuffle lowering on a AMD Jaguar cpu (which is AVX but not AVX2). On this particular target, there is a delay when output data from an execution unit is used as input to another execution unit of a different cluster. For example, There are 6 executions units which are divided into 3 execution clusters of Float(FPM,FPA), Vector Integer (MMXA,MMXB,IMM), and Store

Hi! I fear this may be slightly off-topic but it's both related to IMAP and to dovecot: Is there any good and in paticular realiable program for synching / migrating one IMAP mailbox to another? The most important features for me is that a) no mails are lost / left out silently and b) the porgram is able to sync the complete mailbox including all folders without just giving up in the

Hi, On Sun Jun 12 09:11:13 CDT 2005, Rich Adamson wrote: > > > exten => s,1,Wait(1) > > exten => s,2,GoTo(s,1) > > > > If I'm on the console when a call comes in, it loops through this bit of > > code a bunch of times. I'm guessing I could lengthen the "Wait(1)" time, > > but is there any other way to do this? > > Sure there is,

I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant