Displaying 20 results from an estimated 4000 matches similar to: "Call file & IAX Trunk: Call Failed, Reason 0"
2007 Mar 07
2
Asterisk Auto-dial out
I am using the * auto-dial out feature but don't want to have to specify
a channel (Zap/G2/) to connect to the extension.
Current file I use:
Channel: Zap/G2/12127778866 #<< ==== I have to specify a specific
channel
MaxRetries: 1
RetryTime: 60
WaitTime: 30
#
# Assuming that your outgoing call logic is kept in the
# context called [line1out]
#
Context: line1out
Extension: 7632
2014 Jan 31
2
callfiles.call
hello list,
i have created a callfiles with my asterisk 1.4.43 like:
Channel: SIP/watara/06xxxxxxxx
MaxRetries: 10
RetryTime: 5
WaitTime: 20
Context: mycontext
Extension: s
Priority: 1
extensions.conf
mycontext
exten => s,1,Ringing()
exten => s,n,Playback(hello-world)
exten => s,n,Dial(SIP/105)
exten => s,n,Hangup()
it works with one number how can i do in order to create a
2005 Feb 14
2
Can't run AGI for outbound call
Hi
Just installed Asterisk on a Debian Woody/testing.
I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago).
The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory:
the test.call file looks like this:
#Simple test call script.
#call my
2009 Oct 09
1
${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/155555555
Callerid:Tests <155555555>
MaxRetries: 0
RetryTime:
2003 Dec 22
7
call files
I am after using a web crm system which has a button to then get
asterisk to dial the contact. For this I was looking at call files,
which appear good for the job, I have one small problem with them
though.
1/ file is created
2/ external number is called
3/ the external party answers
4/ the external party now hears ringing as you extension is now being
called - bad!
What I would like to
2006 Mar 28
2
Dial out .call files File permissions??
Hi all,
I've created this test.call file and it is not running outgoing call files:
i've made mv test.call /var/spool/asterisk/outgoing and nothing happens
Channel: SIP/200
MaxRetries: 3
RetryTime: 40
WaitTime: 25
Context: from-internal
Extension: 200
Priority: 1
My asterisk is running with asterisk user. not root user.
Could you help me on ? Could this be a problem of file
2003 Sep 26
3
dialing out with the outgoing queue problem.
Hi,
I have cvs updated all my modules (zapata, libpri, zaptel and asterisk).
I have also read in the archives & seems that no-one has run into this
problem.
What I'm trying to do is simple. Just make and outbound call using the
/var/spool/asterisk/outgoing directory.
I copied /usr/src/asterisk/sample.call and only changed the context &
extension.
I configured my Zap1 to the same
2004 Nov 26
1
How to transfer value to extensions.conf?
Hi, all,
I met a problem for several days, any suggestion is really appreciated!!!
I'd like to do autodial using Asterisk.
For example, I have a file under /var/spool/asterisk/outgoing, which include:
channel: zap/g1/12345
MaxRetries: 0
RetryTime: 60
WaitTime: 20
Context: default
Extension: 2222
Priority: 1
And in my "extensions.conf" file, I have
[default]
exten =>
2003 Oct 08
2
pbx_spool and contexts
When I drop my file into the outgoing folder, the call is completed but
the 'Context' entry is not respected. Instead, it drops into the default
context. It does drop "properly" into the default context and function as
would be expected. I looked through the source but didn't see any reason
it would be completely ignoring the context.
Call file: (where
2006 Jan 05
1
ChanSpy via external application
Hi,
I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface.
This way, I can know the status of my Agent real time.
Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call.
My idea was to, when the user clicks on the Agent, I would Originate a call
2008 Dec 23
2
outging ---asterisk -bug
Hi everyone,
when i use the automated dial out,I found that once the zap answerd,the
contex will be exectued, but i don't hope do it ,i hope when extern phone
answered ,then ,the context will be exectued.
Anyone can help me solve the problem!
the call file is:
Channel: Zap/g0/15015895665
Context: myivr
RetryTime: 60
MaxRetries: 2
Waittime: 60
Extension: 808
Priority: 1
Callerid:
2007 Oct 15
2
About .call files when the congestion is on my side
Hello everyone.
I'm working on an application that needs to automatically send faxes. To
send the faxes I create .call files but the .call files mostly fail
because my lines are always congested within business hours! Is there
any trick I can use to give the end user a better chance at actually
receiving the faxes?
I already tried using the local channel for dialing (so I can put in
2011 Apr 23
2
call files
Hi.
Im having trouble setting variables in channel dialplan and re-using them in
Extension dialplan...
Im using the following call file:
Channel: Local/210332450 at ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context: agents
Extension: 888210332450
Set: __PARTNER=ZonNew-Outbound
Set: NUMBER=210332450
-
In
2007 Mar 07
1
auto dialer
Not able to get the auto dialer part of asterisk to work with the zap
channel. It works great with the sip channel. Here is the call file and
the CLI output
Call File
Channel: ZAP/G1/6144994925
MaxRetries: 3
RetryTime: 40
WaitTime: 2
Context: amaxx
Extension: 36652
Priority: 1
CLI Output
Connected to Asterisk SVN-branch-1.4-r57207 currently running on
VoIP-PBX (pid
2005 Sep 15
2
Caller ID for auto outgoing calls
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic
calls, but I'm having trouble setting the Caller ID for the second half
of the call.
In other words, when we call the first number, we want the Caller ID
set to our number, but then when we connect them to the second number,
we want _their_ number to be the Caller ID.
I've tried the following (and various
2011 Apr 04
1
MeetMe headache
Ok, I've been running applications on 1.4 for quite some time using
meetme to hold a person, while the person on the other end of the call
accepts, etc. I was playing status messages to the calling party using a
context like this:
[status-one-en]
exten => 100,1,Playback(my_status_message)
exten => 100,1,Hangup()
and then creating a call file like this:
Channel: Local/100 at
2007 Feb 07
2
Type of wake-up Call
Hi there,
Is there a way to program asterisk to dial an extension Monday to Friday
at a specific time and then read a specific string? eg: "Kids, go to
the bus stop now, you're about to miss the bus!"
Many thanks,
Pierre
2003 Aug 25
2
SetVar on sample.call
Hi all!!
Does anyone have a short example or even better - a working AGI script that uses "GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses "SetVar"?
Here's what I've tried with no luck so far:
sample.call
=================
Channel: SIP/1000
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Application: Agi
Data: playTasks.agi
Callerid: Nightly Processor
2014 May 15
1
Call file problem, DelayedRetry/retrying spite MaxRetries: 0
I am using Realtime extensions as well, in case that would matter.
Following problem arises from time to time, a call will successfully
terminate:
[May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing
[t at project_init:1] Hangup("SIP/peer-2-00002f7e", "")
[May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init,
t, 1) exited non-zero on
2006 Apr 10
2
HTML / PHP
Has anyone made, or have any simple PHP, or HTML interfaces where by a user
could enter their number and the number they want to call, and have asterisk
bridge the calls?