Op 08-10-12 15:17, Olivier schreef:
2012/10/8 Michel Verbraak mic...@verbraak.org
mailto:mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
Hi,
Any body has an idea on this ? I believe the configuration is correct. Is there
any bug in this version ? Is there any version in 1.8 branch which has it
working ?
Please help.
Regards
Shanavaz.
--- On Sat, 10/6/12, Shanavaz E A shanava...@yahoo.com wrote:
From: Shanavaz E A
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the Idle state to notify the watcher before you
hangup the fxs phone! Such a way if the user forgets to hangup the fxs
phone
Hello
Yes, has a berckeley database, wirh function blackllist
Regards
On Oct 9, 2012 12:51 AM, Joseph syscon...@gmail.com wrote:
Can someone refresh my memory how blocking incoming call works based on
caller ID in Asterisk 1.8?
If I remember correctly in asterisk 1.4 it was possible to block
On 10/09/2012 12:28 AM, Brett Lehrer wrote:
How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into ready-to-send fax image files while the reverse
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
To: asterisk-users@lists.digium.com
Sent: Friday, 5 October, 2012 11:46:48 AM
Subject: Re: [asterisk-users] LDAP Driver and VoiceMail
On 10/04/2012 10:00 PM, Phil Daws wrote:
Hello:
I am investigating the
I'm working on setting up incoming fax reception on our * server. The
majority of faxes come through fine. However each timed a fax comes
in, I get a bunch of this:
WARNING[6616]: res_fax_spandsp.c:416 spandsp_log: WARNING T.30 ECM
carrier not found
Should this be of concern to me? A snip of the
Il 09/10/2012 13:34, Niccolò Belli ha scritto:
Hi,
I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a
remote peer and an fxs phone gets connected and the remote peer hangsup,
then asterisk sends the Idle state to notify the watcher before you
hangup the fxs phone! Such a way
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that
will
be used for local calls. What is the best way to use those lines as
a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
There are lots of
After upgrading to Asterisk 1.8.15.1
I'm constantly getting this error on the command line:
ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get
failed to retrieve iax/provisioning/cache
Can somebody explain what it is and how to fix it?
Since you say this happens
On 10/09/2012 07:40 AM, Steve Underwood wrote:
On 10/09/2012 12:28 AM, Brett Lehrer wrote:
How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into
On 10/09/12 10:55, Richard Mudgett wrote:
After upgrading to Asterisk 1.8.15.1
I'm constantly getting this error on the command line:
ERROR[2499]: iax2-provision.c:266 iax_provision_version: ast_db_get
failed to retrieve iax/provisioning/cache
Can somebody explain what it is and how to fix it?
Excellent. I'll give it a try.
(Now if I just didn't have to wait to get on-site where those lines are
to try it. Too bad there isn't a DAHDI emulator for SIP lines.)
Mitch
On 10/09/2012 10:48 AM, Richard Mudgett wrote:
There are lots of things documented in chan_dahdi.conf.sample. The
On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote:
(Now if I just didn't have to wait to get on-site where those lines
are to try it. Too bad there isn't a DAHDI emulator for SIP lines.)
You can use dynamic DAHDI spans to simulate this on a single box if you
would with
Minor correction below:
On Tue, Oct 09, 2012 at 12:32:44PM -0500, Shaun Ruffell wrote:
On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote:
(Now if I just didn't have to wait to get on-site where those lines
are to try it. Too bad there isn't a DAHDI emulator for SIP lines.)
The Asterisk Development Team has announced the release of libpri 1.4.13.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri
The release of libpri 1.4.13 resolves several issues reported by the
community and would have not been possible without
On 10/09/2012 02:00 PM, Asterisk Development Team wrote:
The Asterisk Development Team has announced the release of libpri 1.4.13.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri
The release of libpri 1.4.13 resolves several issues
Hi
who is responsible for this mailing list? i am not able to post to it.
Br
Adnan
Sent from my iPhone
On 9 okt 2012, at 21:04, Matthew Jordan mjor...@digium.com wrote:
On 10/09/2012 02:00 PM, Asterisk Development Team wrote:
The Asterisk Development Team has announced the release of libpri
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if one
goes down, he can use the other for phone service.
So, I'd like to get a recommendation for a relatively inexpensive router
that can perform this function.
Also, when the failover
On Tue, Oct 9, 2012 at 12:16 PM, Adnan 112linuxstockh...@gmail.com wrote:
Hi
who is responsible for this mailing list? i am not able to post to it.
You just did.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth
Il 09.10.2012 21:24 Mike Diehl ha scritto:
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if
one goes down, he can use the other for phone service.
So, I'd like to get a recommendation for a relatively inexpensive
router that can
Edgewater 4350 or cheaper vigor 2910 dreytech
On Tue, Oct 9, 2012 at 3:24 PM, Mike Diehl mdiehlena...@gmail.com wrote:
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if one
goes down, he can use the other for phone service.
So,
I found that I had to chmod 666 /dev/dahdi/* to allow asterisk to use
the simulation channels. The /dev/dahdi directory seems to be recreated
when dahdi starts.
Here is what I finally came up with that works for me.
system.conf
dynamic=loc,1:0,4,0
fxsks=1-4
dynamic=loc,1:1,4,0
fxoks=5-8
On Tue, Oct 09, 2012 at 03:41:49PM -0500, Mitch Claborn wrote:
I found that I had to chmod 666 /dev/dahdi/* to allow asterisk to
use the simulation channels. The /dev/dahdi directory seems to be
recreated when dahdi starts.
Here is what I finally came up with that works for me.
I am sure Mikrotik routers will do this also, although I have not tried
it.
Niccolò Belli darkba...@linuxsystems.it wrote:
Il 09.10.2012 21:24 Mike Diehl ha scritto:
I hope no one considers this off topic...
I have a phone customer who wants 2 Internet connections so that if
one goes
Here's what I came up with. Works find with the simulated DAHDI dynamic
local channels. I'll find out later in the week how it works with real
hardware.
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1) ;
exten =999,1,Goto(dialpsap,1)
exten
On 10/9/2012 3:52 PM, Niccolò Belli wrote:
http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance
I achieved fallback in less than 10 seconds flushing routing cache and
nat tables with nearly zero false positives (I can do even better but I
prefer having less false disconnections).
Hi,
I am investigating about some SIP redundancy method. I found this article
http://academiccommons.columbia.edu/download/fedora_content/download/ac:109760/CONTENT/cucs-011-04.pdf
and I will try to implement. But, I'd like to ask you, somebody had
implemented some method? Do you have
On 10/08/2012 05:15 PM, Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 11.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users
Il 09.10.2012 23:04 James Sharp ha scritto:
Do you have your phones set for a short register time? Otherwise the
far end might have stale contact information to send incoming calls
back to.
Actually I use the failover only for the nat clients, my pbx has a
public ip on the interface and it
On 10/10/2012, at 9:54 AM, cov...@ccs.covici.com wrote:
I am sure Mikrotik routers will do this also, although I have not tried
it.
Mikrotik can do this but it takes some setup. They are very powerful but what
you are asking is complex and may require the following
- 2 ethernet upstreams or
Hi all,
I am new to Asterisk, and would like to begin by saying that it is an
absolutely fantastic system. Seems incredibly stable, well tested, and easy to
use.
Now, to my question. I am making a mix between a personal ads and a voicemail
service, where I want each user to be able to submit
10.9.0. I'm trying to have a setup where hitting # sends the called
party to the confbridge. I've set GOTO_ON_BLINDXFR:
CLI dialplan show globals
.
GOTO_ON_BLINDXFR=tel-incoming^confbridge^1
(Also tried tel-incoming,confbridge,1 and using | )
but it doesn't work:
Dial(DAHDI/1-1,
I am setting up with meetme a conf with X number of asterisk boxes and
other devices and phones. I am using the l parameter for all devices
being listen only
but I'm not sure thats happening as I am getting some feedback (some
devices are close to each other like 5 feet).
How do I ensure that
Jerry Geis wrote:
Hola,
I am setting up with meetme a conf with X number of asterisk boxes and
other devices and phones. I am using the l parameter for all devices
being listen only
but I'm not sure thats happening as I am getting some feedback (some
devices are close to each other like 5
35 matches
Mail list logo
