Displaying 20 results from an estimated 4000 matches similar to: "spool directories and filename"
2008 Dec 03
3
disable database
Hi,
How do I disabled asterisk to use database and storage voicemail in
directory.
Im getting the below error
[Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to
obtain database object for 'asterisk'!
[Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index:
Failed to obtain database object for 'asterisk'!
[Dec 3 19:09:04]
2012 Aug 01
2
Problem with callfile and CDR
Good afternoon list.
I am experiencing a problem with the CDR and callfiles. What is happening
is this: When generating a call with a callfile, everything works
perfectly, but the CDR is recorded in the table when they answer the call
destination. The field disposition is being recorded correctly, but the
duration field is marked with the ring time and billsec is marked with 0.
This just happens
2013 Oct 31
3
Realtime Call Files
Hi all,
Is there any way of originating calls in future without using call files?
We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we activate
the second server but we lose the call files.
I could have a cronjob on both servers and create callfiles reading
execution time from database, but this involves some other
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
2004 Jul 30
2
Outgoing *-initiated calls from spool directory not working
I'm running:
Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a
i686 running Linux
I've tried placing files (both ending in .call and not) in the correct
format in /var/spool/asterisk/outgoing.... I get _nothing_. No log
messages, nothing on the console, zip. Permissions seem to be correct
on both the files and the directory, as well (* is running as root, for
right
2008 Oct 26
3
hammering imap vmail storage
I've configured asterisk 1.4 to use imap storage for voice-mail and
while I'm happy with it generally speaking it really seem to hammer the
IMAP server. It appear, from the IMAP server logs that it's polling
the imap server every *second* for mailbox updates for the users'
voice-mail folders.
Is it really necessary to do this once a second? Is this tunable
anywhere?
Thanx,
b.
2006 May 24
1
Placing call files in /var/spool/asterisk/outgoing/ does not work
Hello everyone
I'm trying to make asterisk get a call out using the .call system.
The setup is A@H 2.6
This is the content of the file is :
<<<
Channel: Zap/g0/052MYPHONE
MaxRetries: 2
RetryTime: 60
WaitTime: 30
#
# Assuming that your local extensions are kept in the
# context called [extensions]
#
Context: ext-local
Extension: 210
Priority: 1
>>>
I'm
2013 Jun 24
2
Asterisk-11 loop behaviour
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
Snom870 Handsets
We are in the process of moving to an Asterisk based PBX. At the
moment most things work as we wish. However, I have just notices that
when I force a reload using 'amportal a reload' I see this loop start
in 'asterisk -rvvvvvvvvvv':
> Channel Local/s at tc-maint-000002a4;1
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:
2004 Aug 23
1
H323 outgoing calls
Does asterisk support using an H.323 provider for outgoing calls? From
everything I have found, it looks like it does. However, I have had no
success in getting it to work. I would really appreciate if somebody
could give me a hand. I am using the channel that comes with asterisk.
I have also tried using the channel from inaccessnetoworks but have not
had any more success. My provider
2005 Oct 15
2
What would cause a high memory usage in pbx_spool.c ?
Hi,
After only 4 days I have 107472352 bytes in 46007 allocations in file
'pbx_spool.c'
asterisk*CLI> show memory summary
180 bytes in 2 allocations in file 'netsock.c'
12 bytes in 1 allocations in file 'devicestate.c'
2268 bytes in 1 allocations in file 'jitterbuf.c'
8160 bytes in 1 allocations in file
2013 Sep 28
1
problem to get MWI working
Hello,
I am trying to get MWI working after integrating Asterisk with CCM.I have followed the instructions in http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+IntegrationMy problem is that I don't see externnotify's script being called at all in the logs, and not sure if I miss something here!
In Voicemail general I addedpollmailboxes =
2010 Apr 13
1
Merge .csv files
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2 asterisk nodes sharing every configuration file (by using a distributed filesystem) except the master.csv files. If asterisk does not keep master.csv file open between writes, then I can share the master.csv file between both nodes right?If not, then any suggestions to merge both master.csv files?
Thanks in
2011 Mar 15
1
[1.4] Failed callfile doesn't jump to "failed" extension
Hello
For some reason, when dialing out through a call file and the remote
line is busy, Asterisk doesn't jump to the "failed" extension in the
context used by the call file:
====== call file
Channel: Zap/1/5551234
Context: callbacktest
Extension: start
Priority: 1
MaxRetries: 1
====== extension.conf
[callbacktest]
exten => start,1,NoOp(Status is ${DIALSTATUS})
exten =>
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime
instructions on voip-info seem pretty straight forward... just not woking for
me. I've included all of my config files below, and my console output.
Entire console bootup output:
[root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing
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
2009 Sep 02
1
Skype for Asterisk callfile question
Hi list,
To make outgoing calls by skype i would like to have our crm app create
callfiles like we do for normal calls.
If i read the instructions it says this :
---quote---
The syntax for making an outgoing call using Skype for Asterisk is as
follows:
Dial(Skype/[<originator>@]<destination>)
---unquote---
So i create a callfile that looks like this:
---
Channel: SIP/228
2003 Apr 04
2
chan_h323 problems....
I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc.
I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs
pwlib
openh323
gnugk for h.323 gatekeeper
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
2007 Oct 30
6
Newb to puppet - Solaris svcadm question
I''m new to puppet so this might be a foolish question.
I''m using Solaris 10, server is Sparc, my client is x86.
I created a ''file chown'' manifest as a test ...
/etc/puppet/manifests/classes/puppet.pp
class puppet {
file { "/etc/puppet.txt":
owner => "root",
group => "root",