On Thursday 14 December 2006 13:31, cb wrote:
Is there a searchable archive of this list? Did I overlook something
obvious? I can find the archives, but short of downloading all the
monthly gzips and building my own searchable database, it seems my
only other option is to go month by month
On Thursday 16 November 2006 06:44, Conrad Wood wrote:
On Thursday 16 November 2006 06:42, Matthew J. Roth wrote:
As per ManxPower at #asterisk, it is not possible to record a call
dialed from an analog phone connected to the Phone In port of an X100P
because the two ports on the card are
On Wednesday 08 November 2006 13:15, Ken Williams wrote:
I was planning on using a TDM400P with 3 FXO 1 FXS, with the 1 FXS
being used for a fax machine. It now appears that Digium doesn't
support this, are there other manufacturers anyone can recommend that
will support it? Has anyone used
On Monday 23 October 2006 21:45, Angel Heart wrote:
Hi,
Could anyone knows what went wrong with the error below result of
installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf
libsupertone-0.0.2.tar
[snip]
libsupertone-0.0.2/aclocal.m4
[EMAIL PROTECTED] latest]# ./configure
On Wednesday 18 October 2006 05:47, Conrad Wood wrote:
To do something similar, I created a dialplan extension that - if
dialled - creates a file on the server. If dialled again, it removes the
file again.
Then, in the context of the phone I check for existence of that file and
if it exists I
On Tuesday 17 October 2006 17:07, Carla Schroder wrote:
hey all,
I'm getting this warning on the console when I leave a voicemail on my test
server:
[Oct 16 20:56:36] WARNING[3853]: app_voicemail.c:6552 vm_exec: Prefixing
the mailbox with an option is deprecated ('[EMAIL PROTECTED]').
[Oct
On Friday 22 September 2006 15:21, Sean Kennedy wrote:
I just recently purchased some iaxy devices. Being new to this, I
didn't have the iaxyprov tool, so I downloaded the instructions and
attempted to follow them. Below is the problem I ran into.
[EMAIL PROTECTED] src]# svn co
On Friday 01 September 2006 16:32, Ronald Wiplinger wrote:
2 years Asterisk sounds strange, since I can remember there was a bug
with the date a year ago. If you have not upgraded, than this bug is
still in your code. Maybe you just meant no reboot for two years.
That bug was only in one
On Friday 25 August 2006 08:39, existx wrote:
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist
It looks like you have created 2699 in a different context than your
On Wednesday 16 August 2006 10:18, Hadley Rich wrote:
I've just been playing with the STRFTIME dialplan function and am having
trouble getting it to pickup my systems local timezone.
According to show function STRFTIME and voip-info.org all the arguments are
optional and according to voip
Hi all,
I've just been playing with the STRFTIME dialplan function and am having
trouble getting it to pickup my systems local timezone.
According to show function STRFTIME and voip-info.org all the arguments are
optional and according to voip-info.org if you leave them out they will
default
On Friday 11 August 2006 18:26, Wolfgang Paul Rauchholz wrote:
Aug 11 08:00:24 WARNING[2612]: channel.c:2706
ast_channel_make_compatible: No path to translate from
SIP/30-09dfbdb8(4) to SIP/3470075-09e01778(256)
Aug 11 08:00:24 WARNING[2612]: app_dial.c:1595 dial_exec_full: Had to
drop
On Saturday 12 August 2006 14:30, Marco Mouta wrote:
[Aug 12 03:27:35] VERBOSE[26610] logger.c: [format_mp3.so][Aug 12
03:27:35] WARNING[26610] loader.c: missing mod_data for format_mp3.so
What could be wrong?
Looks like you have an old format_mp3 module in your module directory.
Removing
On Monday 07 August 2006 06:36, Chris Hembrow wrote:
I am new to asterisk, and learning as I plod along. Currently, I am
trying to work out how to create a ring group without using AMP.
You should check out the book - 'Asterisk: The Future of Telephony' -
published by O'Reilly it's available
On Saturday 22 July 2006 11:49, Adrian wrote:
Anybody know about (open source with java or C++ ) Softphone support G729
?
At a guess, none because it costs money.
hads
--
http://nicegear.co.nz
New Zealand's VoIP supplier
___
--Bandwidth and
On Friday 21 July 2006 10:39, David R. wrote:
My question is this:
Where can I find good starter documentation(s) for my purposes?
O'Reilly have published a book 'Asterisk: The Future of Telephony' under a
Creative Commons licence. This is usually a good place to start.
You'll find it here;
On Wednesday 05 July 2006 15:10, David Beckerdite wrote:
Is there an archive for this list that can be searched? If so, could
someone tell me where it's located?
http://www.google.co.nz/search?q=site%3Alists.digium.com/pipermail/asterisk-usersie=UTF-8oe=UTF-8
--
The person you rejected
On Wednesday 21 June 2006 18:36, Tyler Retzlaff wrote:
Hello,
I am altering an asterisk configuration and would like to eliminate
the loading of
modules I do not want or do not need at the moment. For example I am
do not
want to use chan_zap (I'm using chan_capi) and don't want to be
On Wednesday 21 June 2006 03:30, Brian Swan wrote:
2. Use fxotune in zaptel-trunk: Find a silent-termination test
number from the phone company and use FXOTune. I couldn't get it to
dial right in order to get silence on the line. You can verify if
it's working correctly by running it
On Saturday 03 June 2006 09:37, Douglas Garstang wrote:
Aaron,
I'm trying to check-in (is that the right term?) the files for the first
time. There's nothing in the repository yet.
http://svnbook.red-bean.com
hads.
___
--Bandwidth and Colocation
On Saturday 03 June 2006 10:05, Douglas Garstang wrote:
[stuff regarding subversion]
http://subversion.tigris.org/servlets/ProjectMailingListList
--
Never try to explain computers to a layman. It's easier to explain
sex to a virgin.
-- Robert Heinlein
(Note, however, that
On Wednesday 31 May 2006 09:08, Julian Lyndon-Smith wrote:
Ok, I must be really stupid here -
I'm playing with ael and svn trunk.
given the following in ael:
context isdn10 {
444601 = {
Answer();
NoOp(${CALLERIDNUM});
Hangup();
On Thursday 18 May 2006 18:35, stoffell wrote:
Aside from being available.. What driver does it use?
Will it be needing bristuff ? (that wouldn't work I guess)
Or will the near future integrate BRI ( and hfc?) drivers in asterisk?
And thus, making bristuff obsolete? (wich means, BRI users
On Thursday 18 May 2006 08:59, Wayne Gemmell wrote:
Does Digium make a quad BRI card? I can't see anything of the sort on their
page but I thought they might call it something else in the States.
They do, but it isn't released yet. Put B410P into google and you will get a
couple of hits.
On Friday 12 May 2006 16:50, Kerry Garrison wrote:
I have written up an guide on how to do bulk provisioning of the Linksys
phones and ATAs.
http://voipspeak.net/index.php?option=com_content
http://voipspeak.net/index.php?option=com_contenttask=viewid=73
task=viewid=73
Thanks for the good
On Thursday 04 May 2006 20:53, Asterisk wrote:
The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
documented is should)
Ie, you cannot use them with intercom or Page features.
Works fine here;
SIPAddHeader(Call-Info:\;answer-after=0)
hads
--
You buttered your bread,
On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote:
Why does Asterisk wait for these two rings? What is it doing meanwhile?
Is it possible to shorten this interval to have an immediate response?
It's most likely waiting on callerid info. If you set usecallerid=no in your
zapata.conf
On Tuesday 25 April 2006 05:50, Mike Garey wrote:
When someone calls into our asterisk server over a PSTN line, dials an
extension and then hangs up, the SIP phone related to the given
extension will ring about 4 or 5 times before asterisk shows that the
channel has been hung up in the
On Wednesday 12 April 2006 18:51, MBIT Technologies wrote:
[regarding the Draytek Minivigor 128]
Any idea where I can get some of these units in Melbourne?
According to Draytek AU they have been discontinued :(
hads
--
Age is important only if you're cheese and wine.
On Wednesday 05 April 2006 11:56, Cory Hawkless wrote:
Does anyone know how to set the distinctive ring on the Linksys SPA941?
Try;
SET(_ALERT_INFO=Classic-1)
hads
--
bureaucrat, n:
A politician who has tenure.
___
--Bandwidth and
On Friday 24 March 2006 05:37, Rich Adamson wrote:
Michael Welter wrote:
I'm seeing quite a few Not Found pages when I google lists.digium.com.
Is anyone else getting this?
Yes, and it apparently has something to do with changes made at the
digium server. Don't have a clue whether anyone
On Friday 24 March 2006 12:53, Larry Alkoff wrote:
What do I have to do to dial an exten - with the dial command in it?
Asterisk isn't recognizing commands in my newly created [context].
There is a really good book available here[1] that will answer this and a lot
of other questions easily and
On Thursday 23 March 2006 02:08, Dr. Michael J. Chudobiak wrote:
I have hit a wall configuring a TDM400, I have set these up before
without issue but today I just can't seem to figure out what I am doing
wrong.
I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't
On Thursday 23 March 2006 09:38, Hadley Rich wrote:
Is anyone else having this problem or am I just going mad?
FWIW I just tried an old X100P on the line and it works correctly.
I don't think I am doing anything wrong in my configuration.
OK, self reply again.
Apologies, yes I was going mad
Hi all,
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
On an incoming call the following is produced in the Asterisk console with
verbose 4
-- Starting simple switch on 'Zap/2-1'
Mar 22
On Wednesday 22 March 2006 16:24, Hadley Rich wrote:
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
On an incoming call the following is produced in the Asterisk console with
verbose 4
On Wednesday 01 March 2006 14:15, mustardman29 wrote:
I am aware of Astlinux and the other embedded Asterisk solutions out there?
Astlinux is nice but the problem is that when I hit a snag and need to
incorporate a patch and what not I cannot do that with Astlinux because I
cannot compile my
On Wednesday 08 June 2005 12:25, Richard Smith wrote:
Would a call coming in on the pstn line be answered by the ATA or just get
passed through to the * server (depending on dialplan) to handle?
So basically, the caller does not get charged until the appropriate
extension hanging of the *
On Wednesday 08 February 2006 14:46, Chris Bagnall wrote:
This is incorrect. Whilst the SPA3000 *can* work this way if you wish, it
doesn't have to.
Apologies, you are correct, there is more than one mode of operation.
hads
--
timesharing, n:
An access method whereby one computer
On Wednesday 18 January 2006 07:21, Sean Cook wrote:
Koopmann, Jan-Peter wrote:
I would like to see if during a call a new voicemail was recorded. I want
to send a SMS to mobile phones if someone recorded a message on our
voicemail system.
I can use VMCOUNT to see if there are new
On Monday 16 January 2006 15:20, Esteban Guana-Jarrin wrote:
We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN
network. We are having some problems with the call quality.
Although we can hear the other person's voice quite clear when making or
receiving a call, we
On Saturday 14 January 2006 13:07, Douglas Garstang wrote:
This therefore means that is a maximum of 9 #include statements that can be
put into extensions.conf. This is a SERIOUS SERIOUS (how many times can I
say it?) limitation. I thought Digium said that Asterisk was supposed to be
On Friday 13 January 2006 15:59, James Harper wrote:
Can anyone recommend a PRI-to-TDMoE device? Does such a thing exist?
Have you seen the Redfone foneBRIDGE? I have no experience of it but it seems
to be what you are after.
HTH
hads
--
I WILL TRY TO RAISE A BETTER CHILD
I WILL TRY TO
On Tuesday 10 January 2006 11:40, Amir Aziz wrote:
I have just installed [EMAIL PROTECTED] version 2.1. It keeps working fine
for couple of days. But after couple of days I start getting the following
error as the Asterisk does not start automatically so I try to start it
with asterisk
On Tuesday 10 January 2006 12:36, Dan Littlejohn wrote:
I have fixed this before, but I cannot for the life of me remember how I
did it.
I have a TDM400P with one fxo module and one fxs module. I setup
Asterisk @Home and everything works fine, except when I try and call
out. If I call out
On Friday 30 December 2005 07:19, Blake Krone wrote:
Hey everyone I have my Asterisk server setup as the DMZ on my Linksys
router. If I use the internal IP as the domain in Xlite clients will
register and work, however, if I use the FQDN for my asterisk server the
clients will not
On Friday 18 November 2005 15:32, Tom Rymes wrote:
Basically, I have 14 after-hours mailboxes that all have different e-
mail addresses. I want this script to parse the mailbox number from
the original command ($2), and then somehow look that up mailbox's
address and send an e-mail. It
On Monday 14 November 2005 16:26, Rich Adamson wrote:
As I recall, the driver for the x100p was called wcfxs (or something
like that), and those driver functions were merged into wctdm about a
year ago. Now, wcfxs is an alias for wctdm.
I've noticed a lot of people referring to wctdm and from
On Friday 28 October 2005 12:06, Richard Smith wrote:
[EMAIL PROTECTED] 1.2.0 beta4 writes to the respective voicemail directory and
when the call is hung-up the .wav file disappears.
Sounds like voicemail.conf is setup to delete the message after it is emailed
to the user.
You may also want
On Friday 28 October 2005 16:22, Eric Bishop wrote:
Does anyone have a full list of places Asterisk puts all config files and
binaries. I need this to be able to fully rollback if I have a failed
upgrade of Asterisk/Zaptel/LibPRI. So far I have:
/etc/zaptel.conf
/etc/asterisk/
Hi,
Since we're doing this...
There is now a New Zealand Asterisk Users Group set up.
There is a wiki and mailing list at http://astug.org.nz both are sparse at the
moment and could do with some input.
If you're in New Zealand (or not) and interested in Asterisk then sign up and
get
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