Folks,
I'm trying to slow down the speed at which Asterisk
decides I've finished typing in an extension for
forwarding voicemail. I've tried using
exten = s,2,DigitTimeout(5);
exten = s,3,ResponseTimeout(5) ;
in my extensions.conf, but it still seems only about 2
seconds (or less!) before
--- Scheda [EMAIL PROTECTED] wrote:
Whenever I try to install [EMAIL PROTECTED], I get this
error at about 43%
There was an error installing
rpmdb-redhat-3.4-0.20050105. This
can indicate media failure, lack of disk space,
and/or hardware
problems. This is a fatal error and your
Hi, all.
We needed to add a few custom menus to the IVR, so got
some recordings done professionally over the past
week. Now, the files (very good sound quality) have
been delivered to us, in .wav format. From googling at
the digium site, I'm seeing that people in the past
have successfully used
--- trixter http://www.0xdecafbad.com;
[EMAIL PROTECTED] wrote:
This script does mp3- wav - gsm, if you omit the
first part it should
work for you. requires mpg123 and sox for wav-gsm
conversion and
mp3-decoder to start from mp3.
will process all .mp3 files in the current
directory..
and all help would be much appreciated!
Thanks!
BeOnIce
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Here is the entire output until it dies:
[EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp
Set to realtime thread
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Not found
(No such file or directory)
Asterisk
Denis wrote:
--- Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu:
beonice wrote:
Ouch ... error while writing audio data: :
Broken
pipe
What are the messages before this?
Matt I think that is something related to mpg123
--- Colin Anderson [EMAIL PROTECTED]
wrote:
Ouch ... error while writing audio data: : Broken
pipe
Did you run make samples from /usr/src/asterisk?
Hmm. I thought I had, but I've attempted this Asterisk
installation so many times that I seem to have skipped
it on this incarnation. I just
--- Matt Riddell [EMAIL PROTECTED] wrote:
Colin Anderson wrote:
Ouch ... error while writing audio data: : Broken
pipe
Did you run make samples from /usr/src/asterisk?
Yeah he has config files.
What is the response you get running mpg123?
I.E. what are the first four lines?
, is what running mpg123 tells me.
Cheers,
BeOnIce
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Asterisk
--- Brian Dingman [EMAIL PROTECTED] wrote:
Put /usr/local/lib in /etc/ld.so.conf then run
ldconfig.
Hmm. I don't understand what that did, but THANKS,
Brian. It seems to have at least got Asterisk to
successfully run. Now I can have fun with configuring
it!
Thanks a bunch!
BeOnIce
Thanks, Brian. I'm now a happy camper!
Cheers,
BeOnIce
--- Brian Dingman [EMAIL PROTECTED] wrote:
It has to do with spandsp and receiving incoming
faxes. This should
probably be updated in the documentation.
On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice
[EMAIL PROTECTED] wrote
of the machine.
HELP.
Thanks,
BeOnIce.
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By the way, in case it matters, I don't have any
digium cards or SIP phones attached to the Asterisk
setup. Right now, it's all IAX, just receiving
incoming calls.
Cheers,
BeOnIce.
--- Chamberland-Larose, Guillaume [EMAIL PROTECTED]
wrote:
If the asterisk process is hung up you should be
able
else encountered
problems with Hangup? Some folks had suggested that
the freeze could be indicative of a hardware problem,
but it doesn't seem likely given that changing Hangup
to SoftHangup seems to remove the problem.
Any help would be much appreciated.
Thanks,
BeOnIce
Mike,
I'm using Asterisk over a wireless LAN (Netgear
something or the other). I use my desktop as the
server and a laptop for remote administration and
testing. So far, I haven't had any major problems that
I would attribute to the wireless connectivity.
Cheers,
BeOnIce
--- Mike Meyer [EMAIL
Folks,
I'm trying to get Asterisk to load my voicemail
configuration from MySQL. I've followed the
instructions at:
http://www.voip-info.org/wiki-Asterisk+voicemail+database
I restarted Asterisk, but no luck: the voicemail.conf
does not get updated. I started with a sample
voicemail.conf that I
--- Matthew Boehm [EMAIL PROTECTED] wrote:
What version of asterisk?
-Matthew
Asterisk CVS-v1-0-12/12/04-15:58:29 built by
[EMAIL PROTECTED] on a i686 running WhiteBox
Enterprise Linux
By the way, I _have_ created an ast_config db and the
content of my ast_config table is:
-
From: beonice [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 11, 2005 10:36 AM
Subject: [Asterisk-Users] Asterisk-MySQL: Not
loading voicemail config
fromMySQL
Folks,
I'm trying to get Asterisk
I've been continuing to experiment with MySQL. I'm
having absolutely no luck getting asterisk to read
voicemail configuration data and mailbox configuration
data from mysql tables instead of from voicemail.conf.
The default Asterisk setup that reads from
voicemail.conf and extensions.conf works
I'm still (doggedly) trying to get asterisk to read my
voicemail configuration from MySQL. I'm using the
stable release of Asterisk, from back in December,
before realtime was included.
If anyone has got it to work, please contact me ...
I've posted details, but everyone who's responded so
far
Lonnie,
If you look at:
http://www.voip-info.org/wiki-Asterisk+RealTime
it says that MySQL _is_ supported.
I don't know whether RealTime PostgreSQL, but I can't
upgrade to RealTime anyway ... I need a stable version
of asterisk, and the current stable version does not
include RealTime. :(
I
did. And now
the core asterisk can see my voicemail configuration
in MySQL just fine. I must have originally retrieved a
buggy version of the stable asterisk. :)
Thanks, everyone, for all your help!
Cheers,
Maya
--- Kevin P. Fleming [EMAIL PROTECTED]
wrote:
beonice wrote:
I don't know
in the extconfig.conf file so it can find it.
beonice ([EMAIL PROTECTED]) wrote:
I've been continuing to experiment with MySQL. I'm
having absolutely no luck getting asterisk to read
voicemail configuration data and mailbox
configuration
data from mysql tables instead of from
voicemail.conf
troubleshooting tips? This is frustrating! I
_have_ asked a question via my VoicePulse Connect web
account, but I don't know when they will respond.
Thanks,
BeOnIce.
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to comment out the #include!
Has anyone run into this weird behaviour? I am not
convinced as to where the problem is, given that the
message still seems to indicate somehow that the
interaction with voicepulse is the problem.
Thanks,
BeOnIce.
--- beonice [EMAIL PROTECTED] wrote:
I've been using my
Folks,
I've been running asterisk successfully using the
extensions.conf and voicemail.conf.
Now that I've got asterisk happily looking up MySQL
tables for the VM configuration, I decided to try out
the contributed script
/usr/src/asterisk/contrib/scripts/retrieve_extensions_from_mysql.pl
I
--- Andrew Thompson [EMAIL PROTECTED] wrote:
--- snip ---
The only thing that seems out of place to me is your
#include in
[main_vp_context]. It looks to me like you intend
for the s, #, t, and i
extensions to be in [main_vp_context]. The way you
layed out this
example, that's not
Robert,
Thanks for the suggestion. I also came to the same
conclusion, but Asterisk wasn't showing any errors.
Then I decided to go back to basics, so I backed up my
extensions.conf and used the sample one provided by
voicepulse voila, it worked.
The culprit? Me. I'd commented out the
this line is so important?
I'm assuming that incoming calls in other countries
will have different patterns for the incoming dialed
number. How does asterisk handle that? Can we add the
explanation to that question into the FAQ somewhere?
Thanks again, everyone!
Cheers,
Maya
--- beonice [EMAIL
--- Andrew Thompson [EMAIL PROTECTED] wrote:
... snip ...
Back to your original post...
As of yesterday, though, when I have this format,
asterisk won't accept incoming calls. It barfs
with
the message:
Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
socket_read: Rejected
--- Andrew Thompson [EMAIL PROTECTED] wrote:
beonice wrote:
The culprit? Me. I'd commented out the line:
exten = _NXXNXX,1,Background(welcome) ;
which is apparently a critical one. I was under
the
impression that
exten = s,1,Answer
Will s be traveled if a call arrives
-Users] Anyone having
trouble with
VoicePulse Connect?
On Feb 17, 2005, at 4:32 PM, Andrew Thompson
wrote:
beonice wrote:
The culprit? Me. I'd commented out the line:
exten = _NXXNXX,1,Background(welcome) ;
which is apparently a
critical one. I was under the impression
--- beonice [EMAIL PROTECTED] wrote:
... snipped ... this thread is getting long ...
Hmm. I guess that would be one way to do it.
But we still have the issue of what happens when
calls
come in from DIDs in other countries. How are our
colleagues in Europe and Asia handling
Robert, thank you very much for that informative
write-up. Of course, I now have more questions. The
first is really basic. I thought extension meant
something the caller dials _after_ reaching asterisk.
How come incoming DIDs have to be handled as if they
are extensions?
More questions follow:
--- Martijn van Oosterhout [EMAIL PROTECTED]
wrote:
--- snipped my quote of what 's' is --
I guess it
implies that calls coming from DIDs have digits
associated with them.
Correct. On ISDN lines, E1, T1 and related digital
protocols, details
such as CallerID, Dialled Number, CLI
Andrew, thanks for the explanation ... see more
questions below. :)
--- Andrew Thompson [EMAIL PROTECTED] wrote:
beonice wrote:
--- snipped some ---
I guess the fundamental question is why is a call
coming in from a DID any different? And, of
course,
does a call coming
It looks like it's breaking at the iax.conf file. Have
you set up your iax.conf with the registration info
your service provider gave you? It should look
something like this:
register = iaxid:[EMAIL PROTECTED]
So, in my case, I have a line that says
register = myid:[EMAIL PROTECTED]
where myid,
Thanks, Robert. Yes, I _finally_ figured out why I
need multiple extension contexts. I'm now one happy
camper.
Thanks again,
Maya
--- Robert Hajime Lanning
[EMAIL PROTECTED] wrote:
quote who=beonice
Robert, thank you very much for that informative
write-up. Of course, I now have more
Dean,
I'd be very interested in helping with this effort.
I've worked with both SGML and XML in the past (I used
to work at SoftQuad in Toronto, one of the original
providers of SGML and HTML tools), and have written
several DTDs, both for SGML and XML.
I think it would be fun to work on an XML
Try this:
In the extensions_custom.conf file set:
[custom-myapp]
exten = s,1,SayDigits(1234)
exten = s,2,Hangup()
You're telling it to go to priority s,
(exten = 3,1,Goto(custom-myapp,s,1)
but in the custom-myapp context, you have priority 3
instead.
Hope that helps.
Maya
--- Chris
I'll let someone else speak to the missing .conf
files.
If you could post your extensions.conf and
extensions_additional.conf, it would be easier to help
you debug this. The suggestion from timebandit
_should_ have worked given your original post.
Personally, for learning my way around
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call
--- Michael Loftis [EMAIL PROTECTED] wrote:
--On Monday, February 28, 2005 08:46 -0800 beonice
[EMAIL PROTECTED]
wrote:
-- snipped --
When _I_ dial
either DID, I get exactly the same behaviour that
I
have specified (the call is answered, and then I
play
my own welcome
--- Colin Anderson [EMAIL PROTECTED]
wrote:
Has anyone had production experience using * w/
MySQL Blobs to store sound
files? The application I am working on requires
all user data resides in a
database. I am currently reading/writing the
files to disk via a phpagi
scripts but I would
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2005-03-07 at 13:30 -0800, beonice wrote:
... snip ...
Would it help to split the db off to a separate
server
(that should reduce the CPU load on the asterisk
server)?
Any other alternatives? Anyone verified whether
Folks,
I want to build a custom IVR for my setup. I've got it
working (well, the bells and whistles are not there
yet, but the basic stuff works) using AGI, but I'm
worried about how well this will scale.
I've seen references to FastAGI, and presumably this
will be more efficient.
Question,
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Mon, 2005-03-07 at 14:21 -0800, beonice wrote:
--- snip ---
Would it make more sense to write a custom
application
in C instead, designing it to work sort-of like
the
built-in app_voicemail.c and others? I do know C
--- J P Edmund [EMAIL PROTECTED] wrote:
I have been searching all over for the answer on all
sources online and
have come to the conclusion that it must be
rudimentary or I am asking
the wrong question.
I cannot figure out how to configure the box to set
the from address
to a correct
--- Steven Frazier [EMAIL PROTECTED] wrote:
I am trying to SetCallerID to a variable I have
defined. This obviously is
wrong. It actually sets the caller ID to
$NEWCALLERID. I have search
through the examples on wiki but wasn't able to find
something similar to
see what I was doing
Hello, all.
Google searches reveal that a couple of users have
used Libretel port-of-call service via FWD using
Asterisk. I don't have a FWD account. I don't have a
SIP phone (no SIP setup at all, in fact).
I just purchased a DID in Brazil from Libretel and
discovered that they assume that I
Folks, I'm having trouble with my voicepulse numbers.
Over the past week, incoming calls have been very
slow to be answered, but they seem fine while the call
is in progress. When the caller hangs up, asterisk
takes a while (over 2 minutes in some cases). This
system does not make outgoing
--- Tim Burt [EMAIL PROTECTED] wrote:
I was having problems with voicepulse about a week
or two ago...
Incoming calls would fail, and one incoming call,
would block all outgoing
calls.
Then one day, the DTMF tones stopped working.
I could call into Asterisk, but I could not
Folks,
I'm hoping someone has already run into this ... the
only other complaint I've seen is here:
http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html
and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.
I've been using
--- Tim Connolly [EMAIL PROTECTED] wrote:
Is NAT=yes on, are you behind a firewall? Give us
some connectivity details.
Usually when you see maximum retries, its because
you have one-way
communications with the far end for some reason. Are
you setting externip
statically?
To answer your
--- Tim Connolly [EMAIL PROTECTED] wrote:
Is NAT=yes on, are you behind a firewall? Give us
some connectivity details.
Usually when you see maximum retries, its because
you have one-way
communications with the far end for some reason.
Are
you setting externip
statically?
I accidentally posted this on the biz list,
but it probably belongs here.
---
Folks,
Has anyone successfully got a Libretel DID forwarded
to a Voicepulse Connect number for incoming calls?
I have signed up for an overseas DID with Libretel. My
main VOIP number is a
--- Michael D Schelin [EMAIL PROTECTED] wrote:
Hi all, What would cause the greetings not to play.
The u command is
supposed to play the unavailable greeting. It
doesn't work. with this
setup. Maybe I'm missing something. The voice
prompts play well. What
do you think? Thanks
exten
Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context.
My problem is that my overseas DID through Libretel
seems to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I
--- Jeff Heath [EMAIL PROTECTED] wrote:
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to
voicemail from one of the
phones.
This is a test system that I setup about a
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server| NAT
Hello, all.
I'm tired of having to drive out to the colocation
facility each time my dedicated asterisk server craps
out, just to press the button to do a hard reboot.
(I'm running 1.05 stable at present, no telephony
hardware, as this is mainly a system that receives
calls, no dial-out ability
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of beonice
I'm tired of having to drive out to the colocation
facility each time my
dedicated asterisk server craps out, just to press
the button to do a hard
reboot.
(I'm running 1.05 stable at present, no telephony
hardware
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Hi
On Thu, Jun 23, 2005 at 05:52:44PM -0700, beonice
wrote:
Okay, so what makes more sense:
1) a remote management card that will let me
actually log in to the machine to monitor it as
well
as to reboot it
vs.
2) a remote
Jason, when I've seen this, it's usually meant that I
was logged in as a user who didn't have the right
permissions. :)
Check your notes to figure out which user you gave the
permission to (root, maybe?) and log in as that user,
then try to connect using asterisk -r.
Hope that helps,
Maya
---
--- [EMAIL PROTECTED] wrote:
Actually this is on an enterprise system - I have
never seen more
than one. Just checked now and there were several
calls up and only
one ps.
On Jul 1, 2005, at 8:33 PM, Luki wrote:
Do not know why, but have noticed redhat = 1, and
debian = many
Hello, all.
This is rather annoying. I know that this _used_ to
work in my production system.
I'm running Asterisk 1.0.5 and am using IAX
exclusively.
Used to be, when someone called in from a
caller-id-blocked number, the notification email said:
From an unknown caller, and the message
Folks,
I've been trying to handle the problem where
blocked callerids appear as coming from
asterisk asterisk
on the email notification, and the message
envelope simply doesn't say anything (does not
actually play the vm-unknown message).
So, following the tip provided by several
previous
on their
incoming voicemail notifications and voicemail
envelopes!
Cheers,
Maya
--- beonice [EMAIL PROTECTED] wrote:
Folks,
I've been trying to handle the problem where
blocked callerids appear as coming from
asterisk asterisk
on the email notification, and the message
envelope simply doesn't say
--- Dan Journo [EMAIL PROTECTED] wrote:
Hi Beonice,
Just told i was dealing with the same problem.
[incoming]
exten = _!,1,GotoIf($[${CALLERID} =
unknown]?2:5)
exten = _!,2,Set(CALLERID(name)=Withheld Number)
exten = _!,3,Set(CALLERID(number)=00)
exten = _!,4,Goto(8)
exten = _!,5
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