2009/3/8 Marco marcota...@libero.it
Hi List,
I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years
on my lab test setup and I appreciate it. Moreover the global quantity of
fax handled by this setup is not very high.
I'll be involved in a more complex system for a
2009/3/8 Sven Geggus use...@fuchsschwanzdomain.de
Gavin Henry gavin.he...@gmail.com wrote:
Just transfer them to your meetme extension after you've called them.
Hm, how would I do this? Until now call switching usually ended for me when
the call has been established.
I'm using a SIP
Hi,
I have several GXP2000 phones which used to work fine with Asterisk 1.2.
However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from
a GXP2000, it gets dropped after 20 seconds exactly.
I have early dial enabled on the GXP2000 and pedantic=yes on the server. If
I
Hi All,
Asterisk 1.4.12 on CentOS 5
Yesterday and today I got the following warnings in /var/log/asterisk/messages:
WARNING[2066] chan_sip.c: Got authentication request (401) on unknown REGISTER
to 'sip:acco...@sip.voipuser.org;tag=d8f15e1f30efddd35168b07dba9d540e.3922'
The corresponding bits
On Fri, Mar 6, 2009 at 11:59 PM, Tiago Durante tiagodura...@gmail.com wrote:
On Fri, Mar 6, 2009 at 10:39 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
Danny Nicholas wrote:
The log files themselves are not in color. It would be a style sheet change
on the GUI.
-Original
Paul Hales wrote:
Noojeeclick?
http://www.noojee.com.au/Page/NoojeeClick
Thanks for that. Not heard of NoojeeClick before. Their site is not
responding right now but the Firefox add-on page is up. when I get
chance I will try it out.
https://addons.mozilla.org/en-US/firefox/addon/8510
I
Hi All,
For my setup, i am using a macro to dial a certain extension not just a
simple Dial(SIP/exten).
I would like to setup a ringgroup, for now what i only found is by
simply dialing like this Dial(SIP/exten1SIP/extenSIP/exten3) but i
cant use since i'm using a macro, is there a way i can
Hi:
How can I see the communication between hylafax and iaxmodem and the console of
them?I only can see the console of asterisk.It shows: 'IAX2/iaxmodem-2 is
ringing' when I dial the fax number.
and nothing else.I can't receive fax.
I installed asterisk 1.4.18 and iaxmodem-1.2.0 and
On 23 Feb 2009, at 15:13, Dean Collins wrote:
Asterisk/Skype update available here -
http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/
…. It’s definitely an update that updates absolutely nothing J, more
news at 11 :P
John Todd and I discussed this at some length on the
fateme fatah wrote:
Hi:
How can I see the communication between hylafax and iaxmodem and the
console of them?I only can see the console of asterisk.It shows:
'IAX2/iaxmodem-2 is ringing' when I dial the fax number.
cd /var/spool/hylafax/log
tail -f your.log.name.here
Doug
--
Ben
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370,
my problem is that when arrived an external call I don't view (on my
internal phone) the phone number but I have the number extension that is
configured on MV-370.
The MV-370 configuration is:
Mobile to Lan Table :
0 *
Hello,
I need to execute an agi in php.
I have that:
== Using SIP RTP CoS mark 5
-- Executing [0170725...@mnupprx1:1] Answer(SIP/33179977999-b6c18478,
) in new stack
-- Executing [0170725...@mnupprx1:2] GotoIf(SIP/33179977999-b6c18478,
0?6:3)) in new stack
-- Goto
The message indicates that DEADAGI will not work and that you should use
AGI instead. Are you sure PHP is installed on your machine and functioning
properly (from $, php a2billing.php works)?
_
From: asterisk-users-boun...@lists.digium.com
Hello list,
I have this strange problem whenever I try to make an ael reload from the
Asterisk CLI. The command gives the following result and crashes:
voip-1*CLI ael reload
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
r...@voip-1:/etc/asterisk#
As far
I have the same thing with AGI in the dialplan
And php is install
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de
You didnt say whether a2billing.php works from the shell. Is it 755
permissioned?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Monday, March 09, 2009 8:53 AM
To: 'Asterisk Users Mailing List -
On Sun, Mar 8, 2009 at 9:44 PM, Mark Michelson mmichel...@digium.com wrote:
Caution: One shortcoming of queue member penalties is that they are not
taken into account if a queue member of a low penalty does not answer a
call. Say for instance that the queue application determines that there
I have all permissioned
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : lundi 9 mars 2009 15:07
À :
I am sorry it work !
In fact, I had mistakes in my config
Sorry
And thank you for answering
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
6 mar 2009 kl. 13.36 skrev Mikel Lindsaar:
Hi all,
Is there any way to make use of the SIP making progress messages?
I find that about the time the SIP peer says making progress is the
time the other end actually starts to ring, or is busy etc.
Before that time, I want to generate an in
Hi Gordon, thank you for your answer.
It's not mandatory to use an external box to handle the PRI. I was
thinking to use a Patton device instead of a TE120P just because I would
like to be able to switch to T38 in the near future or if working with
inband faxes will reveal problems.
I'm open to
Marco Signorini wrote:
Analyzing your answers, seems that fax handling is still today
problematic with IAXModem and Hylafax... or I'm wrong?
I would strongly disagree with this statement.
I'm running 3 systems. All 3 are PRI/iaxmodem/HylaFAX+. I may see a
failure maybe once every other
Marco Signorini wrote:
Analyzing your answers, seems that fax handling is still today
problematic with IAXModem and Hylafax... or I'm wrong?
A single server that I administer, receiving 12,000 pages and sending
1,000 pages daily would seem to contradict your conclusions.
Thanks,
Lee.
Marco Signorini wrote:
Hi Gordon, thank you for your answer.
It's not mandatory to use an external box to handle the PRI. I was
thinking to use a Patton device instead of a TE120P just because I would
like to be able to switch to T38 in the near future or if working with
inband faxes will
On Friday 06 March 2009 06:26:13 am Robert Broyles wrote:
Great backports! :-)
This should really be merged into 1.4.
This would violate the release policy for 1.4.
--
Tilghman
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On Mon, 9 Mar 2009, Danny Nicholas wrote:
The message ?indicates? that DEADAGI will not work and that you should use
AGI instead.
From Wikipedia, deprecated means:
In computer software standards and documentation, the term deprecation is
applied to software features that are superseded and
Thanks Doug and Lee,
your testimonials are changing my opinion :-)
Can you provide some details about your setup? What PRI solution are you
using? And what version of Asterisk, IAXModem, SpanDSP?
Thank you and best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
On Friday 06 March 2009 09:39:30 am Johann Steinwendtner wrote:
Sorry, that I wasn't clear enough. The logfiles contains escape codes +
the colour codes.
e.g.:
[Feb 12 13:38:30] VERBOSE[19816] logger.c: == Registered custom function
'ESC[1;36;40mSQL_ESCESC[0;37;40m' [Feb 12 13:38:30]
Hi,
I'm trying to install spandsp from source in a Debian Lenny system.
I did :
cd /usr/src
wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre5.tgz
tar xvf spandsp-0.0.6pre5.tgz
cd spandsp-0.0.6
./configure
make
make install
When doing this, spandsp warns me that librairies are
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with cdr_odbc.
As we know, after answering each call a cdr event is raised which is saved
in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some
information,
On Friday 06 March 2009 03:02:30 pm Randy Paries wrote:
On Tue, Feb 3, 2009 at 2:58 PM, Jose P. Espinal j...@slackware-es.com
wrote:
Hello List,
I have been working on a little PHP software that uses AMI's
UpdateConfig command in order to modify some of it's config files.
I was
Hi Steve,
I was waiting for your answer :-P
I started to use your SpanDSP library since some years ago but,
unfortunately, my experience was only related to lab or personal use
and/or systems with PSTN or BRI cards and low fax volume where it's
impossible to have valid statistics.
I read the
On Mon, Mar 9, 2009 at 11:13 AM, Olivier oza-4...@myamail.com wrote:
Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in my
opinion, spandsp libriaries have not been found.
Maybe, I should have typed something like (as suggested
Marco Signorini wrote:
Thanks Doug and Lee,
your testimonials are changing my opinion :-)
Can you provide some details about your setup? What PRI solution are you
using? And what version of Asterisk, IAXModem, SpanDSP?
Main fax server:
Mandriva 2008.1
Kernel 2.6.24.5 (Compiled for
Hi all,
I am having trouble setting the signalling method for the B410P using
DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or
'bri_net' - but it doesn't mind having 'pri_cpe' etc.
ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method
'bri_net'
Dahdi -
2009/3/9 Sasa s...@shoponweb.it
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370,
my problem is that when arrived an external call I don't view (on my
internal phone) the phone number but I have the number extension that is
...
..now what parameter can I modify
Thanks for proving what I said. In *, deprecated means you will be dead
when you go to the next release.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday, March 09, 2009 11:08 AM
To:
On Mon, Mar 09, 2009 at 05:36:48PM -, Andrew Thomas wrote:
Hi all,
I am having trouble setting the signalling method for the B410P using
DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or
'bri_net'
Those only work as of Asterisk 1.6.0 .
- but it doesn't mind having
Hi
What it's the result of execute
strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI Telephony'
It's LibPri install before of Dahdi package?
JL.
El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió:
Hi all,
I am having trouble setting the signalling method for the
si n?cessaire.
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Message: 2
Date: Mon, 9 Mar 2009 16:08:08 +0100
From: Olle E
I just upgraded a very old Asterisk installation to the last 1.2.31 I
can find in Asterisk.org site. Now for some reason my IAX clients
cannot connect to the server. I can do a iax2 show peer iaxmodem1 and
I get this:
* Name : iaxmodem1
Secret : Set
Context :
Tilghman Lesher wrote:
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with cdr_odbc.
As we know, after answering each call a cdr event is raised which is saved
in cdr_csv and cdr_odbc. but here my point is on cdr_odbc.
Did your iax.conf get overwritten with the upgrade?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Carlos Chavez cur...@telecomabmex.com wrote:
I just upgraded a very old Asterisk installation to the last 1.2.31 I
can find in Asterisk.org site. Now for some reason
No, it is the same one. I have tried creating iax.conf from scratch
with the same results.
On Mon, 2009-03-09 at 13:32 -0500, Tim Nelson wrote:
Did your iax.conf get overwritten with the upgrade?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Carlos
2009/3/9 James Sneeringer jsnee...@gmail.com
On Mon, Mar 9, 2009 at 11:13 AM, Olivier oza-4...@myamail.com wrote:
Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in
my
opinion, spandsp libriaries have not been found.
Maybe, I should have typed something like (as
2009/2/26 Olivier oza-4...@myamail.com
I must add I tried spandsp0.0.6xxx as a warning message advised me to do so
(using 0.0.4 would be ok for me but current trunk doesn't allow this
anymore, it seems).
2009/2/26 Olivier oza-4...@myamail.com
Hi,
With 0.0.6pre3:
# ./build.sh
CMake
On Mon, 9 Mar 2009, Danny Nicholas top posted:
The message indicates that DEADAGI will not work and that you should
use AGI instead.
Steve Edwards wrote:
From Wikipedia, deprecated means:
In computer software standards and documentation, the term deprecation
is applied to software
On Sun, 8 Mar 2009, Carlos Chavez wrote:
I just upgraded a very old Asterisk installation to the last 1.2.31 I
can find in Asterisk.org site. Now for some reason my IAX clients
cannot connect to the server. I can do a iax2 show peer iaxmodem1 and
I get this:
Yes, it's broken in
On Mon, Mar 9, 2009 at 3:16 PM, James Sneeringer jsnee...@gmail.com wrote:
If you are using dynamic queues with Local channels (as described in
doc/queues-with-callback-members.txt in the Asterisk source), you can
also optionally implement this functionality directly in the dialplan.
This
What you proved was that you will eventually diss whatever I write. Yep, I
do have a short memory; * is only about 20% of my job function.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent:
Hi All,
Have any way to send one announcement to calling party and other
different announcement to called party when the call are briged ?
Bruno
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asterisk-users mailing list
Hi,
I have a customer running a 120 second long WAV file on their MoH. The
problem is that it's always starting from the beginning, so people being put
on hold, talked to, put on hold again, etc always hear the first 10-15
seconds.
Is there a way to have Asterisk MoH remember where it
Hey guys,
I'm having a really huge problem, it seems like Asterisk is locking my
licenses of g729 after being used.
For example, 10 people make calls using this codec. Then I can see the
channels and the codecs being used, cool. But then when they hang up
the call the codes are still there, as
According to the documentation I've read, MOH will always start at the start
of the file. You could possibly put all of the on-hold folks into a
conference room, but a more transparent option would be to create
staggered versions of the wav file and add random=yes to moh.conf
1.wav = original
nik600 wrote:
On Mon, Mar 9, 2009 at 3:16 PM, James Sneeringer jsnee...@gmail.com wrote:
If you are using dynamic queues with Local channels (as described in
doc/queues-with-callback-members.txt in the Asterisk source), you can
also optionally implement this functionality directly in the
Running an earlier version of Asterisk (1.2), we were using Hints to show
busy extensions on other (SNOM) phones.
When we went to version 1.4 they stopped working, using the same syntax.
(Copied and pasted)
Does anyone have any tips or clues?
Is the exact location in the file critical?
The hints may have been moved to a different context. On Polycom phones,
the hint either has to be in the default context or specified in the
directory (1...@somecontext).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hello list!
I'm looking for someone who is local to Atlanta and is proficient in coding
(PHP, MySQL, TCP/IP) and has knowledge of Asterisk. Knowledge in any CRM
technologies, screen-pops integration, large call volume is also helpful and
puts you right in my face.
I do not mean to disrespect
Mike wrote:
Hi,
I have a customer running a 120 second long WAV file on their MoH. The
problem is that it's always starting from the beginning, so people being
put on hold, talked to, put on hold again, etc always hear the first
10-15 seconds.
Is there a way to have
Mark Michelson wrote:
Mike wrote:
Hi,
I have a customer running a 120 second long WAV file on their MoH. The
problem is that it's always starting from the beginning, so people being
put on hold, talked to, put on hold again, etc always hear the first
10-15 seconds.
Is there a
Anthony Francis wrote:
Tilghman Lesher wrote:
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with cdr_odbc.
As we know, after answering each call a cdr event is raised which is saved
in cdr_csv and cdr_odbc. but
On Monday 09 March 2009 01:28:49 pm Anthony Francis wrote:
Tilghman Lesher wrote:
On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote:
hi,
I'm working with asterisk on a project and I found a problem with
cdr_odbc. As we know, after answering each call a cdr event is raised
which is
It should be posted on the biz list if you care. There are also many
forums, include a Digium owned one.
On Mon, Mar 9, 2009 at 3:46 PM, Sean McMaster sean.mcmas...@msn.com wrote:
Hello list!
I'm looking for someone who is local to Atlanta and is proficient in coding
(PHP, MySQL, TCP/IP)
Great, thanks! I should learn to check Mantis before posting these kind of
questions, my bad.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Monday, March 09, 2009 15:59
To:
Give www.asterisk-jobs.com a try too if you want J it's free.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean McMaster
Sent: Monday, March 09, 2009 3:47 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Job in
To get busy state for a sip channel in 1.4 it appears the peer/friend
must have a call-limit.
Steve
On 3/9/09, Cary Fitch ca...@usawide.net wrote:
Running an earlier version of Asterisk (1.2), we were using Hints to show
busy extensions on other (SNOM) phones.
When we went to version 1.4
According to voip-info.org, the call-limit is mandatory to make hints work
as of 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Davies
Sent: Monday, March 09, 2009 4:07 PM
To: Asterisk Users
On Sat, Mar 7, 2009 at 2:20 PM, Johann Steinwendtner
steinwendt...@gmx.netwrote:
John Todd wrote:
Just a suggestion: have you tried more recent versions of Asterisk
with IAX2? I'm uncertain what version you're using, and if it's
1.2.4, that's getting to be quite old and the problems that
Thanks to all for the hints about Hints. Got them working. Shoulda Read the
Fine Manual.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Monday, March 09, 2009 4:12 PM
To: 'Asterisk
On Mon, 9 Mar 2009, Danny Nicholas top posted, veering far off topic:
What you proved was that you will eventually diss whatever I write.
According to Wikipedia:
to dis, African American Vernacular English slang meaning to
disrespect
I mean no disrespect and I don't target
On Mon, Mar 9, 2009 at 8:39 PM, Mark Michelson mmichel...@digium.com wrote:
The reason that the member always appears to be not in use is that local
channels are optimized away once they are bridged to their real destination.
The
result of this is that since the channel does not exist
On Mon, Mar 9, 2009 at 3:17 PM, Tiago Durante tiagodura...@gmail.com wrote:
Hey guys,
I'm having a really huge problem, it seems like Asterisk is locking my
licenses of g729 after being used.
For example, 10 people make calls using this codec. Then I can see the
channels and the codecs
Hi,
When we use svn branches-1.4 such as:
# svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4
# svn checkout http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4
how to write the others such as dahdi-linux and dahdi-tools?
Regards,
Zen
On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote:
Thanks, i've tested and it works (1.4.23.1).
Just 2 questions:
1) this approach seems to be an hack and not the implementation of a
feature is it really used in corporate solutions?
2) using queue show 001 i can't see the
Mark Michelson wrote:
Remco Barendse wrote:
On Fri, 6 Mar 2009, Klaus Darilion wrote:
Updating to 1.4 branch solved the issue. Thanks.
Pity that they still didn't release a new version that works properly.
We can't afford to release a new version every time we fix a bug. That's just
not
Darrick Hartman wrote:
I know the call parking feature changed in 1.4.23.1 to fix some serious
issues. I'm seeing a major change though which I find disturbing.
A person parks a call by transferring it to the parking position (700).
When the timeout value is reached, the call is NOT
Remco Barendse wrote:
And how do i get that fix? :) Do i need to build asterisk from SVN, if yes
how do i get the right version?
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
Since you appear to be trying to run from the 1.6.0 branch, then you would want
to grab the latest
Leif Madsen wrote:
Darrick Hartman wrote:
I know the call parking feature changed in 1.4.23.1 to fix some serious
issues. I'm seeing a major change though which I find disturbing.
A person parks a call by transferring it to the parking position (700).
When the timeout value is reached, the
thank you Dear doug
But,I don't have any file in /var/spool/hylafax/log directory.
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