Displaying 20 results from an estimated 10000 matches similar to: "NoOp does not seem to be printing messages on the console..."
2010 Dec 22
5
* 1.8: cannot load g729 free codec (on 1.4 it worked!)
pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so
Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so
Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed.
[Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module
'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key.
[Dec 22 15:52:45] WARNING[4491]:
2007 Mar 16
4
proposal: a new mailing list for asterisk 1.4, why not?
Hi all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
2006 Apr 26
4
Excessive Asterisk delay to answer on ZAP inbound call
Hi,
I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P
(12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I
connected an analog phone in parallel to make a test:
__________Asterisk fxo
---- line -----|
-----------------Analog phone
The analog phone rings immediately when calling, while asterisk shows
the message
2006 May 29
3
TDM2400P with echo canceller not working
Hi,
I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a
TDM2400P with echo canceller. I installed the card but no echo
cancellation is being made...seems like the echo canceller module does
not work, infact the software cancellation is working.
My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but
no echotraining parameter which gives a warning.
I found
2004 Dec 21
2
SOHO PBX using asterisk
Hi,
I'd like to build a personal PBX connecting 4 or 5 analogic phones with a
ADSL line and I'd like to know what is the right card I need
I visited digium site and I think TDM400 could be the right choice but I
cannot understand how it works...I think it has 4 slots where 4 modules
(FXS or FXO) can be inserted. How many cards do I need to connect my ADSL
line to 5 phones? I think I
2006 Dec 06
3
Asterisk freezes when DNS not working: a BUG??
Hi,
I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers
registrations: Asterisk freezes when it cannot (re-)register with VoIP
provider (registration timeout). The problem is related to DNS names
resolution: if DNS server is very slow to respond Asterisk stops every
activity (no zap or restart commands on CLI). The bad news is VoIP
providers usually do not give their IP
2006 Mar 15
3
how to show called name on calling polycom display
Hi,
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display instead of his /her
extensions as I do with the caller name on the called polycom.
Is it possible? If yes, how?
TIA
Giorgio Incantalupo
2008 Jul 21
1
queue members randomly become paused after upgrade to Asterisk 1.4
Hi all,
I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems
that sometimes some phones become paused and cannot receive calls
anymore. I tried to set autopause = no in every section of my
queues.conf but nothing changes....
Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or
there is a particular reason for this behaviour?
Thank you.
Giorgio.
2005 Jul 02
1
Is it possible to setup group voicemail in Asterisk?
Hello there,
I'm a new Asterisk user and I wonder if it is possible to associate a
voicemail box with a group of users, i.e., a single recorded message is
sent to everyone in that group. If so, where can I find more
information about that?
Thanks in advance,
Leo Burd
2006 Oct 16
4
Remote UNIX connection, Remote UNIX disconnected displayed every second
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
TIA
Giorgio Incantalupo
2006 Jan 27
2
WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
Hi,
I'm using asterisk 1.2.1.
Is there anybody out there who knows what this warning means?
*WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer:
image 5004 udptl t38*
Google does not help at all.
TIA
Giorgio Incantalupo
2006 Mar 15
5
how to show called name on calling polycomdisplay
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
To my knowledge it is not posible. I don't even think a SIP standard is
available for this.
This 'feature' along with changing CallerID Display after a call has
been answered is something missing from the RFC.
>
2005 Aug 08
2
URGENT: Problems with PHP AGI...
Hello everyone,
I'm having all sorts of problems with my PHP AGI scripts... Basically,
my scripts run fine from the command line and don't do anything well
called from Asterisk. Here are my questions:
a) Does Asterisk require PHP CLI or CGI? From the command line, my
script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi)
b) How to debug my script? According
2007 Jan 22
2
tdm400p not working with brazilian lines
Hi,
I'm installing an Asterisk box with a TDM2400P in Brazil. I can make
analog phones work while lines are not working. Since I do not know
anything about brazilian lines, is there anybody who can tell me what is
wrong/missing in my conf files (below)?
TIA
Giorgio
_zaptel.conf:_
fxoks=9-16
fxsks=17-24
defaultzone=br
loadzone=br*
*
_zapata.conf:_
context = inbound_zap
echocancel = 128
2007 Dec 12
4
TDM400 hangup issue in China
Afternoon,
I was hoping someone could point me in the right direction. I have an
asterisk PBX deployed in China using a TDM400P based card. The incoming
calls are being picked up correctly, but are not being hung up. I
suspect that this might be an issue with the signaling that has been
selected.
If anyone here has deployed asterisk in china using an analog card, it
would be a great help
2003 Apr 23
5
Call Monitoring
Hi,
Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes?
Thanks
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2005 Jul 05
4
How to prevent log files from eating my hard drive?
Hello there,
Somehow, Asterisk log files are consuming all the space that I have in
my hard disk... They've already eaten 14GB and are still hungry!! What
shall I do? I'm not even logging anything in verbose mode!!
Help really appreciated!!
Best,
Leo
2006 Mar 29
3
FOP flash panel: how to reload config files when running
Hi,
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
TIA
Giorgio Incantalupo
2006 May 25
2
connecting asterisk to hylafax via t38modem: is it possible?
Hi,
I'm trying to use Hylafax without a modem. Is it possible to use
t38modem to make Hylafax send and receive fax via Asterisk?
If yes, how? I'm searching on internet but still haven't found anything
useful.
TIA
GIorgio Incantalupo
2005 Oct 17
1
module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format
Hi,
I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel)
using gcc 4.0.2.
Compilation does not give me errors so after a 'make install' I try to
load zaptel module with insmod but the following error arise:
*insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format*
Is there anybody who can help me??
TIA
Giorgio
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