Displaying 20 results from an estimated 70000 matches similar to: "No subject"
2005 Mar 21
2
Ext matching problems
Hello everyone...
I'm trying to get up a testing pbx installation. Following instructions
of what've read from the handbook and from asterisk's wiki, I wrote the
dial plan as follows:
[general]
;
;
static = yes
;[globals]
;
[default]
;
exten => 0,1,Answer()
exten => 0,2,Playback(fcopba1)
exten => 0,3,Hangup()
exten => *0,1,Answer()
exten => *0,2,Record(fcopba1:gsm)
2006 Nov 08
0
Warning: "Channel does not have a CDR" when doing ForkCDR
Gang,
I'm having this error pop up when I do a ForkCDR, and I'm not sure how
to get around it. Here are a few log lines:
Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing
ForkCDR("Zap/49-1", "") in new stack
Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a
CDR
The scenario occurs like this:
I use a .call file to generate a call on
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 Mar 25
2
WaitExten question
I'm a bit confused about how WaitExten works. I assumed that when it
returns 0, the next priority in the extension would be executed, but
that doesn't seem to be the case. When I get to WaitExten and enter
extension 8, it plays the message, then Waits another 10 seconds and
times out.
[local]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten =>
2006 May 16
0
Need help with Dial M option and destination context
I would appreciate hearing from anyone who has figured this one out.
Here's the scenario:
I have a context wherein I give the called party the option to dial the
digit 9. If he does so, he is transferred a la this extension entry:
exten => 9,1,Playback(pls-hold-while-try)
exten => 9,n,Noop(Attempting to bridge to ${agentext})
exten =>
2010 Aug 27
0
Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload => format_mp3.so
preload => codec_ulaw.so
preload => format_pcm.so
My extensions.conf
2010 Aug 27
0
Asterisk 1.6 Displaying BackGround() in call trace but no audio is heard from caller
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
---------- Forwarded message ----------
From: Joe Wood <schmoe at gmail.com>
Date: Thu, Aug 26, 2010 at 6:58 PM
Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw
file using BackGround()
2003 Dec 10
0
A solution to "free line" notification
Barton Hodges wrote:
> I've been messing around with a "free line" notification
> where an extension is dialed for a second when a line becomes
> available. I can't seem to get the "h" extension to continue
> when the local party hangs up. I've seen references to other
> people having the same problem in the list archives, and the
> solution
2007 Jul 25
1
Dialtone when automatically picking up.
I'm in the process of setting up a 'phone tree', and are running into
some problems. My goal is for users to dial a phone number, the
asterisk system picks it up, plays the greeting, and users can type
whatever they want into the system.
What actually happens is users dial the phone number, asterisk picks up
and additionally goes off-hook on another line, plays the greeting and
2006 Nov 17
1
Extension Response Slow
Here is my Extensions.conf file (Default Context). When an
individual calling in dials the extension, the response time seems
very slow. It doesn't immediately go to the next step, but hangs out
for a few seconds (silence)... Suggestions?
Thanks in advance... /pj
[default]
exten => _XX.,1,Wait,2 ; Wait a second, just for fun
exten => _XX.,n,Answer
2011 Mar 23
2
using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten =>
2008 Jan 26
3
GotoIf() on Auto-Attendant
Hello all,
I'm planning to create a simple Auto-Attendant (IVR Menu) for my home PBX
yet all callers from incoming (trunk) calls must only press the extension
numbers from the [analog-ext] else will play the "pbx-invalid". How do you
do that using the GotoIf() (or probably using the other applications) but
will check if the numbers entered belongs to a specific context?
Also, how
2007 Dec 16
1
Newbie question: how to proxy the *real* caller-id on find-me/follow-me
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
gateway.
The issue is that my cellphone shows my PBX's number, not the original
calling
2008 Nov 06
0
Asterisk trunking
Hello !
I am experiencing some problems with Asterisk trunking, this is the scenario:
There are 3 servers, a DID server provider (VOIP provider) which
delegates us a bunch of DID numbers to our asterisk server number one
(I will call it AA), from which I route the calls to Asterisk server
number 2 (I will call it BB), which then terminate on phone handsets.
The trouble is, that I probably
2006 Mar 03
4
really need help with outgoing calls..PSTN errors
I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is
2008 Apr 03
1
Hearing "transfer" during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word "transfer", I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
2007 Jul 30
0
Trouble getting sound from a call
Having some issues with getting sound from a call.
I have 4 systems. 3 main systems which handle calls for our 3 locations.
The 4th system is the central voice mail system. When an inbound call
gets passed to someones voice mail its done with an IAX2 connection. The
same happens after hours when we have our night mode set. If you dial
the main number after hours you are passed straight to the
2006 Dec 22
2
Determining invalid extensions.
Hi all,
I'm trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn't see that say, extension 600 is invalid, it just goes back to the
beginning of the callplan and repeats. If I enter a single digit, it
works perfectly. Anyone have any
2007 Oct 18
1
IAX2: Calls answered before extension is tested?
[Sorry if this arrives more than once. I have sent this twice and it
never arrived, despite other messages getting to the list O.K.]
-----------
Hello,
I would like an incoming caller to be able to choose from the menu
options in my extension.conf below. Once They have dialled the
appropriate digit, * should call two extensions simultaneously: one SIP
phone on this * server, and one over a
2009 Jul 21
1
Dialplan step that I do not have
I have a dialplan that looks like this:
[dorecord]
exten => _*99XX,1,Verbose(2,Doing custom record)
exten => _*99XX,n,Answer()
exten => _*99XX,n,Verbose(2,Doing custom record - before wait)
exten => _*99XX,n,Wait(0.5)
exten => _*99XX,n,Verbose(2,Doing custom record - before record)
exten => _*99XX,n,Record(/tmp/prompt${EXTEN:3}.gsm)
exten => _*99XX,n,Verbose(2,Doing custom