Displaying 20 results from an estimated 200 matches similar to: "Extension Response Slow"
2006 Nov 14
0
(no subject)
Here's a question maybe someone can help me with:
My extension looks like this:
exten => 1006,1,MP3Player,http://audio-mp3.ibiblio.org:8000/wcpe.mp3
When I try this extension, the following output appears in the CLI:
Nov 13 12:47:51 NOTICE[8422]: app_mp3.c:111 timed_read: Poll timed out/errored out with 0
I should mention that mpg123 is installed, however the server we are using for this
2005 Sep 06
1
Queue AgentCallBackLogin
Hi All,
I'm having trouble setting up a queue: I'm using AgentCallBackLogin to
login in the queue, but:
1 - When an agent answer the call and another call arrive his phone
rings again.
2 - When no there are no one answer the queue the system goes to
voicemail of agent 1234
I'm using asterisk-1.2.0-beta1.
My configuration is below,
Any ideas?
Many thanks,
Joao Antunes
2013 Dec 06
1
Paging in waves.
I've been working on writing a subroutine to page groups of phones at once
and I'm having some difficulty.
My goal is to have a user call an extension, I record the page they wish to
play, I then page out that recorded file to the phones in groups.
[sub-masspage]
exten => s,1,NoOP
same => n,Answer
same => n,Set(filename=$PAGE)
same => n,Wait(1)
same =>
2006 May 25
0
Re: Implementing Paging on the Linksys SPA9XX phones (working)
I came up with this a few days ago, mostly used the wiki examples,
didn't have time to post on the wiki yet, maybe one of you guys with a
few minutes can throw it up there, really, I forgot my logon.
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom
The agi script didn't work for me, wouldn't call the active hint
extensions, even though they were there, no
2006 Jan 24
0
Safari problems w/ Effect.Appear and other effects
Prototype version: 1.4.0
Effects version: 1.51
I''ve recently been working alot with Scriptaculous and Prototype and
I''ve got some good looking results in FF and IE. However, Safari is
killing me! I would love for someone to look at this, and tell me if
its my design or the libraries.
One example:
http://www.tankdb.com/src/report_demo.php
1. Checkout the above page
2.
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
2007 Feb 05
1
How to access environment variable?
How can I access an environmental variable in Asterisk 1.2.5?
It should be possible according to:
http://www.voip-info.org/wiki/view/Asterisk+variables
which says:
Environment Variables
You may access unix environment variables using the syntax:
${ENV(foo)}
${ENV(ASTERISK_PROMPT)}: the current Asterisk CLI prompt.
${ENV(RECORDED_FILE)}: the filename of the last file saved by the Record
2013 Apr 29
1
Asterisk 11.3.0 - Mask for new file not correct
Hello,
I'm facing a rights issue on with Asterisk 11.3.0 running on CentOS release 5.8. Asterisk process is running with asterisk since it is define in asterisk.conf as following:
runuser = asterisk
rungroup = asterisk
You can see asterisk proccess here:
ps aux |egrep 'python|asterisk'
root 11581 0.0 0.1 65940 600 ? S Apr17 0:00 /bin/sh /usr/sbin/safe_asterisk
2016 Jun 07
2
Want to detect sound
<!DOCTYPE html>
<html><head>
<meta charset="UTF-8">
</head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["
2003 Dec 18
4
SIP / X-ten Softphone
I know this has been covered more times than to mention and this is
where I got most of my info from... But I am having issues with this. I
can't seem to get the phone to register with *. This is being tested on
a internal network right now.
Here is the setup -
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context
2015 Oct 09
0
Asterisk 11.20.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.20.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.20.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2004 Jan 19
3
Getting correct CDR info
I'd like to know how everyone else is going about getting correct CDR
information for calls. Right now I notice that if a call come in and gets
parked the CDR info doesn't how the correct info on who picked that call up,
also when someone transfer a call there isn't a new record being made so the
duration of the call shows up for who received the call and transferred it.
I started
2013 May 01
0
asterisk-users Digest, Vol 105, Issue 39
*I'm trying to build an application that provides statistics of
calls*>* and call recording. Someone told me this could be done out of
band*>* with a SPAN (?) port that would replicate SIP and media
packets to a*>* separate NIC without having to actually pass the
real-calls thru*>* asterisk. It was explained that this SPAN port
would in the SBC*>* would replicate data
2015 Oct 09
0
Asterisk 13.6.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.6.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New
2004 May 04
2
Can Asterisk support R2 signaling
Hi All:
I'm a newbee to Asterisk. I currently working on a project and want to know
if Asterisk does support R2 Signaling.
Thanks
Begra8fl
>From: asterisk-users-request@lists.digium.com
>Reply-To: asterisk-users@lists.digium.com
>To: asterisk-users@lists.digium.com
>Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs
>Date: Tue, 04 May 2004 13:32:00 -0500
>
>Send
2006 May 12
3
VoiceMail application: "j" option not working as I supposed
I've the following dialplan.
exten => _XX,hint,SIP/${EXTEN}
exten => _XX,1,Dial(SIP/${EXTEN},10,j)
exten => _XX,2,VoiceMail(${EXTEN}@default,u|j)
exten => _XX,3,Hangup()
exten => _XX,102,Goto(110)
exten => _XX,103,Playback(pbx-invalid)
exten => _XX,104,Hangup()
exten => _XX,110,VoiceMail(${EXTEN}@default,b|j)
exten => _XX,111,Hangup()
exten =>
2007 Jun 07
3
getting at ${CALLERIDNUM}
Hi all --
I'm having awesome fun with Asterisk & voicepulse connect together.
So cool.
I'm trying to have the caller id read back to me. Do I need to do
something to have this sent across in the sip.conf? Or is there
something I need to do somewhere to enable the reading of this data?
Thank you!
Matt
Here is my extensions.conf
exten => _XX.,1,Answer()
exten
2006 Dec 22
2
Determining invalid extensions.
Hi all,
I'm trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn't see that say, extension 600 is invalid, it just goes back to the
beginning of the callplan and repeats. If I enter a single digit, it
works perfectly. Anyone have any
2005 Aug 28
1
DIALSTATUS for Originate
Hi all,
I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER as in case of command DIAL when used from the dial plan. Can some one guide me how to get the vaue of
2006 May 09
0
How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels
Is there a way to distinguish, in the answer of the Dial command,
when a channel is not available (for example, an unregistered but
valid SIP user) v/s when the dialed channel is inexistent, even
when it matches an extension?
For example, I've the following simple dial plan:
exten => _XX,1,Dial(SIP/${EXTEN},10,)
exten => _XX,2,GotoIf(DIALSTATUS = CHANUNAVAIL?4:3)
exten =>