Displaying 20 results from an estimated 2000 matches similar to: "Applications, dialplan not loading"
2003 Jun 13
1
strace shows that files are not accessed
strace on file access in asterisk shows that * is not even attempting to
access the voice files.
If I *manually* load app_playback.so, app_macro.so, and then
pbx_config.so, I they will load and I get a dialplan. Ok, that's a
problem -- autoconf is clearly not working, or there's some other
related issue.
So I try to use the demo and do "dial 500". This should connect and
2003 Jun 15
5
.gsm files
Hi guys,
Being a true Linux geek, I've never been too much into sounds or sound
files other than a few .mp3 songs I got. My question is pretty
straightforward and simple. I see that the music format of choice for
asterisk is .gsm. What can I use to listen to files in .gsm format and
what is the most effective way of recording files into .gsm format?
The last part of the question is
2003 Jul 01
2
Today's Message from linphone; update on Khpone and SJPhone and X-Lite
Today's "frustrated programmer" award goes to Linphone, which has the
following debug output:
> (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer!
I get this message when I connect to linphone using a softphone, or when
I try to use linphone to connect to asterisk and listen to an
announcement. I suspect that
2003 Jun 24
2
Asterisk ALSA module not working
Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The
module chan_alsa.so won't load even if the oss module, chan_oss.so,
isn't loaded. There are no error messages.
I've been chasing ALSA/Asterisk/client problems in one form or another
for some time now. In previous versions of Asterisk and ALSA -- i.e.,
last week -- I could load either chan_oss.so or
2003 Jun 30
3
Connections, but no voice paths except by console
I have a software-only PBX set up. I can register various softphones and
they will call each other -- but I've never succeeded in getting any
voice routed from any of the softphones. Only the console will transmit
audio.
I am writing to ask if I have missed some obvious step in configuring
the system.
Conditions:
(1) Softphones running on the same machine as the PBX: Only Kphone seems
2003 Jun 13
5
Disabled echo canceller because of tone (rx)
Does anyone know what this means? It is in DMESG, and we have
people complaining about echo.
Disabled echo canceller because of tone (rx)
John
2003 Jun 15
1
Re: Application, Dialplan not loading
strace does show that that modules.conf loads:
> open("/etc/asterisk/modules.conf", O_RDONLY) = 8
And that I do get some of the channels loading, e.g., the modem channel:
> open("/usr/local/lib/asterisk/modules/chan_modem.so", O_RDONLY) = 8
And if I load the apps via "load app_playback.so",
>
2003 Jul 15
5
Text to Speech - Someone needs to do this
Why hasn't someone found 50 people who sound alike, put them in sound
studios and record the 10,000 most commonly used words. You would all
differnent forms of the 1,000 most words, i.e. leading, trailing, question
etc.
You can synthesize the other 0.05% when you run into them. With hard drives
so big, processors so fast and EXT3 that can handle 30,000+ files in a
single directory that
2003 Jun 24
1
Working Clients for Linux?
All the clients that I'm aware of for IP telephony have drawbacks. Some
won't work at all.
KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to
dial tones during the middle of the call, so the demo that * comes with
can't be run. Kphone (3.1, the latest) also has a habit of crashing if
you do something even mildly stressful, such as hang up while Kphone is
2013 Mar 04
2
Asterisk 11 - How to trim the number of modules to minimum ?
Hi,
I've got a brand new Asterisk 11 setup for which I would like to keep the
number of loaded modules to a minimum.
My goal is to this setup in a pure SIP environment, for switching incoming
calls to outgoing tSIP trunks.
When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an
incoming SIP call with a Playback app.
When I leave autoload=no in /etc/asterisk/modules.conf, it
2009 May 20
2
Manager ExtensionState function
Hi,
I am trying to get the extension status (weather it has dialed
outgoing call via SIP or IAX2), using the following piece of code
however it always returns -1 on all the extensions (valid/invalid).
Am i missing something ? Any help.
Thanks
-----------------------------------
#!/usr/bin/perl
use Asterisk::Manager;
use lib './lib', '../lib';
$|++;
my $astman = new
2006 Feb 01
2
Dundi key Problem
I am getting the following message when trying to lookup up a number via
Dundi:
Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key
'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for
'00:a0:c9:55:91:89'!
I have created keys on each box with "astgenkey -n
office.pbx.bluegrass.net" using the host name for each box of course.
I
2004 Sep 16
3
Creating conference calls from within Astman.
Dear All,
I have a requirement to 'originate' a number of calls to various external
users from within a conference room, so that the end users does not pay for
the call.
I know that within Astman I can define an extension and then originate the
call from that extension. Can I define a conference room (how would I
configure that on astman? What channel would it use?) and then generate a
2007 Feb 01
2
make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed.
hi all
i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0
any suggestions ?
make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect'
Generating input for menuselect ...
menuselect/menuselect --check-deps menuselect.makeopts
Generating embedded module rules ...
[CC]
2006 Apr 29
2
Codec G729 no longer works.
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from Digium.
This is the error message from asterisk -vvg:
[app_playback.so] => (Sound File
2012 Mar 01
1
using AMI and Telnet to place calls
Hello,
I am using a perl script to pull call info from a DB and place calls via
telnet and AMI, all on local machine of course. My problem is that I
need to capture any response from the carier, such as this taht appears
in the CLI:
[Mar 1 12:55:50] == Using SIP RTP CoS mark 5
[Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available"
back from xxx.xxx.xxx.xxx:5060
[Mar
2013 May 11
1
AMI Originate issue
Hi,
I'm getting an issue while executing AMI Originate.
I'm getting "extension does not exists" on Originate's Response, and on the
other hand Asterisk CLI say "fwrite() returned error: Broken pipe"
Please suggest me what is wrong.
Muhammad Faheem
### my originate code block ...
2005 Feb 20
10
HELP NEEDED! - Asterisk GUI
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing my GUI using;
http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl
I am trying to get the menu options in my GUI to work but to no avail.
Currently my parameters are set to;
Asterisk Install Directory:
2011 Jun 16
1
Web based call back
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.
Any suggestions?
CK
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