Displaying 20 results from an estimated 900 matches similar to: "Background Playback fails"
2010 Jun 12
1
Problem launching Cursed mountain
Hello
I installed the game "Cursed Mountain", with no problems, at the end of the installation it asked for launching the game, and the game ran successfully. Later I wanted to launch it again, but now it gets stuck after the logo intro.
The terminal output is:
Code:
---------------------------------------------------
KTM
---------------------------------------------------
Version:
2008 Nov 06
2
Variable Scope Question
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and want to be able to transfer calls as if they were all on
the same phone system. Each company has 4
2006 Apr 05
5
Dial Plan Logic Problem
Hi
I can't for the life of me work out why this is not
working. When in the campon contect if you hit a DTMF
key 2 you get moved to the exten => 2 defined in the
mainmenu context not the exten => 2 defined in the
campon context. What is wrong? The same happens if you
hit key 1.
[campon]
exten => _*1XXX,1,Answer
exten => _*1XXX,2,SetCallerID(${CALLERIDNUM})
exten =>
2006 Nov 19
2
WaitExten only reading 1 digit.
I am trying to setup an interactive menu where a caller hits the main
menu and can then dial an extension. As far as I can tell the
"Waitexten" app is failing to read 3 digits and just reading the first
and then announcing that it is invalid since all extensions are 3 digits.
How do I make Waitexten wait for 3 digits?
I have setup the extension "100" for users to reach the
2005 Jul 23
2
(cause 66 - Channel not implemented) -- IAX?
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new
2006 Jan 06
2
Incoming PSTN Calls - Stumped
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that?s happening (and I?m very stumped
with this)
.I think my contexts are definately the reason that I
can?t interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to
2003 Nov 05
2
Need info on Gastman/Astman
Has anyone used Gastman/Astman successfully?
I have it up and running (Gastman win32), but have a problem with the
creation of end stations on the map. I'm not sure of the format of the
extension to use when creating a end station icon.
Services like Conference bridge and Musichonhold seem to work ok (I use
555@mainmenu and 666@mainmenu) for the Icon extensions.
IAX softphone seems to work
2006 Nov 15
1
simple mainmenu ivr tones not recognized
I'm trying to setup a VERY simple mainmenu ivr but can't seem to get the
tones to be recognized during the background( ) the playback and background
files play, but asterisk doesn't do anything when I start pushing keys -
I've tried it from softphones and pstn line phones
Can anyone tell me what I'm doing wrong? Required contexts Exentions.conf
below
[from-broadvoice]
2004 Oct 05
2
Long pause between menus
I have set up an auto attendant and all is working but I am bothered by
a long pause when switching between menus. This pause is between 5 and
7 seconds and is quite annoying.
Is there anyway to address this.
One other thing I find interesting is that when I move from the main
menu to the sub menu the delay is there but when I move from the sub
menu to the main menu the delay is not there.
2004 Aug 11
2
Autoattendant Configuration
Hi,
At my house, I have two POTS lines. Both are connected to my * server
on a TDM400P card. As an example, say the phone numbers are
(919)555-1212 and (919)555-1213. I also have four SIP extensions, an
ATA with a fax machine, and a DID coming in from an ITSP.
I have an autoattendant configured that talks and allows users to
forward to the extension they choose, but my family doesn't like
2003 Apr 30
2
first few seconds of greeting cut-off
When a person calls into the Asterisk voicemail or auto attendant, the first
second or two are cut-off. This happens with custom prompts I have created
(with or without 1 or 2 second delays) and with the default prompts that
come with Asterisk.
Does anyone have a solution to this problem?
I'm running the current CVS. My default menu config is:
[mainmenu]
;
; We start with what to do when a
2005 Mar 08
1
Dial() out and offer a menu system
Hello all!
I'd like my * to call out to somebody and offer the called party
a menu system. Everything should just be as if the called party
had initiated the call themselves.
This is my try:
exten => 100,1,Dial(Modem/g1:0555321)
exten => 100,2,Goto(mainmenu,s,1)
This doesn't really work, because the Dial cmd blocks further
execution until the called party hangs up.
Next try:
2006 Feb 13
1
Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent
Here is some dialog from the Console:
-- Starting simple switch on 'Zap/13-1'
Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring
Begin)...
-- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack
-- Goto (mainmenu,s,1)
-- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support")
in new stack
2005 Sep 21
1
Call getting disconnected in queue
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started investigating and found that that happens when the call gets
transferred to an agent who is making an
2009 Mar 29
2
h exten no getting run ...
Asterisk 1.4 r181990
given the dialplan snippet below, can anyone tell me why the h exten is
not being run ?
============================================================================
console output:
[Mar 29 10:33:49] -- Executing [s at questionnaire-menu:1]
Set("Zap/1-1", "TIMEOUT(digit)=3") in new stack
[Mar 29 10:33:49] -- Digit timeout set to 3
[Mar 29
2007 Oct 24
2
Help with loop counting?
Hi
I have a situation where I want to be able to count how many times a
caller goes round a loop of "Please hold...", "please continue to hold".
I have found an example on voip-info but I can't get it to work. Not
sure if I've got some syntax wrong somewhere? All that happens at the
moment, is I hit is the playback of "som-debug" at 9999. Any ideas would
2004 Aug 03
2
SPA-3000 as a regular Asterisk FXO device?
My SPA-3000 finally arrived and I'm trying to get the FXO port on it to
work as if it was a X100P card as far as Asterisk is concerned.
I have Asterisk dialing out over the SPA-3000 FXO port no problem.
The issue I'm having problems with is having the SPA-3000 automatically
forward all incoming PSTN calls to the Asterisk "mainmenu" context (or
ext I guess).
Currently the
2012 Oct 12
2
Recommendation for extension mapping on inbound T1 line
Converting this customer from a MiTel system to asterisk. Discovered
that the inbound calls from the T1 are going to extension 366. (This
was mapped in the MiTel for some arcane purpose.) The dial plan I am
currently using is shown below. When loading the dial plan, I get this
warning:
WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for
an extension is strongly
2006 Jan 05
1
Incoming PSTN Calls
Hi all,
I am having difficulty getting incoming PSTN calls working. I have set
up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register => username:password@sip.blueface.ie/2093
; To receive incoming calls specify this block and
2004 Jan 07
1
Call Rollover
Have a question about implementing Call Rollover with my current
extensions.conf configuration.
[macro-stdexten]
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail
w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten =>