?
2013/6/4 Ron Wheeler rwhee...@artifact-software.com
Well, at least you are making progress.
What is the error in the debug log?
Ron
On 03/06/2013 8:03 PM, Olivier CALVANO wrote:
grrr no in asterisk -d i have no error, but when i start normaly
asterisk i have :
[Jun 4 02:01:45
2013/5/7 Matthew Jordan mjor...@digium.com
On 05/06/2013 05:54 PM, Olivier wrote:
Hi,
Before trying to script res-memcached installation (see res_memcached
https://github.com/drivefast/asterisk-res_memcached), I banged into
this on a fresh 11.3.0 setup:
snip
My questions
2013/5/7 Jason Parker jpar...@digium.com
On 05/07/2013 05:13 AM, Olivier wrote:
2013/5/7 Matthew Jordan mjor...@digium.com
2. It appears as if you're running a modified version of Asterisk, in
which case all bets are off. This works fine on the Linux build agents,
which is what we use
...@gmail.com
You the couple opensips + asterisk will help you. Opensips loadbalance
module is your friend.
Sent from my iPhone
On Apr 25, 2013, at 11:44 AM, Olivier oza_4...@yahoo.fr wrote:
Hello,
I've been given the task to study what would a good way to load balance
SIP trafic
Hi,
What are the main differences between Aastra SIP phones 6730i and 6750i
series ?
Aastra corporate web site mentions :
The Aastra 6730i Series offers exceptional features and flexibility in an
open-standard enterprise grade IP telephone for one
The Aastra 6750i Series offers features and
Hi,
2013/4/19 Olivier oza_4...@yahoo.fr
Hello,
I've just realized that several phones display both caller name and number
while ringing but when on call, caller name is not displayed anymore.
Could you recommend a sip phone that still displays caller name during
phone call ?
Regards
I've
Hi,
Before trying to script res-memcached installation (see
res_memcachedhttps://github.com/drivefast/asterisk-res_memcached),
I banged into this on a fresh 11.3.0 setup:
1. When run for the first time bootstrap.sh displays a non-blocking error.
# sh -x bootstrap.sh
+ uname -sr
+ MY_AC_VER=
+
Le 5 mai 2013 12:19, Marie Fischer ma...@vtl.ee a écrit :
On 04.05.2013, at 20:20, Olivier oza_4...@yahoo.fr wrote:
Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
from time to time, we get so-called simplex / one-way audio calls,
where one party cannot hear the other. The only
Le 2 mai 2013 13:23, Marie Fischer ma...@vtl.ee a écrit :
Hello everybody,
from time to time, we get so-called simplex / one-way audio calls, where
one party cannot hear the other. The only thing in common is that is does
happen with calls via SIP trunk, not ISDN and not internal calls.
Hello,
I've been given the task to study what would a good way to load balance SIP
trafic.
The prospective setup is :
- call centers sending outbound SIP trafic (no inbound) from SIP devices
(with public fixed IP address),
- a couple of outbound SIP trunks to which trafic from call centers is to
Hello,
I've just realized that several phones display both caller name and number
while ringing but when on call, caller name is not displayed anymore.
Could you recommend a sip phone that still displays caller name during
phone call ?
Regards
--
If I correctly understood your question, the only way I can think of to
force extension 102 to stop ringing (because of extension 101 rejecting
incoming call) is to pass Asterisk a soft hangup command (using CLI or AMI)
with appropriate ringing channel identification.
Hop this helps.
2013/4/15
2013/4/10 Satish Barot satish4aster...@gmail.com
On Tue, Apr 9, 2013 at 7:56 PM, Olivier oza_4...@yahoo.fr wrote:
Hello,
What about having a new DB_IFEXISTS function in Asterisk ?
It would accept two arguments : one being family/Key just as DB function,
and the other being a default
Hello,
What about having a new DB_IFEXISTS function in Asterisk ?
It would accept two arguments : one being family/Key just as DB function,
and the other being a default value which would be returned if family/key
value does not exist.
Thoughts ?
Regards
--
Hello,
Many times, I need to test in a lab Asterisk servers before sending them to
customer locations.
I'm currently having trouble to test SIP trunks without touching SIP
configuration.
So, how should I change my testing lab so that I could now test SIP trunks
without modifying Asterisk server
2013/4/8 Johan Wilfer li...@jttech.se
2013-04-08 16:36, Olivier skrev:
Hello,
Many times, I need to test in a lab Asterisk servers before sending them
to customer locations.
I'm currently having trouble to test SIP trunks without touching SIP
configuration.
So, how should I change my
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
check out the endbeforehexten option in cdr.conf
this needs to set to yes
Julian
Unfortunately, this doesn't help.
Let's drop the hangup handler at the moment, and focus on the saving to
file part.
Then my issue is I can't update CDR value
)})}
In the actual configuration file, the value in the Master.csv mapping
should be on a single line.
cdr_manager
On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote:
2013/3/29 Julian Lyndon-Smith aster...@dotr.com
check out the endbeforehexten option in cdr.conf
this needs to set to yes
Julian
How would you qualify it ?
A feature ? A bug ?
Could you find a work around ?
2013/3/29 Mitch Claborn mitch...@claborn.net
I have seen that behavior also.
Mitch
On 03/28/2013 06:56 PM, Olivier wrote:
Hello,
I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one
effect.
It works.
You mean it does work in asterisk 11 ?
Which CDR variables did you then play with ?
I know it does, as I use hangup handlers all the time. Much
better than the h extension ;)
I fully agree !
Julian
On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote:
Thanks but I
Hello,
I'm using Hanhup Handlers in a testing asterisk 11 system.
Within one such handler, I'm setting CDR values.
To me, it seems those changed CDR values are not saved in CDR back-end.
Can you confirm ?
Regards
--
_
--
2013/3/26 Richard Mudgett rmudg...@digium.com
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such
as
Asterish and Tech cause (see function HANGUP_CAUSE
Hello,
I'm reading this in my log files:
[Mar 25 12:01:23] WARNING[1593] sig_pri.c: Span 1: Got SETUP with duplicate
call ptr (0x8e3b998). Dropping call.
[Mar 25 13:21:40] WARNING[1593] sig_pri.c: Span 1: Got SETUP with duplicate
call ptr (0x8e3b998). Dropping call.
[Mar 26 10:20:54]
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such as
Asterish and Tech cause (see function HANGUP_CAUSE).
I want to have my custom hangup-handler be run only once and when the
second channel hangs up.
At the
2013/3/25 Matthew Jordan mjor...@digium.com
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such as
Asterish and Tech cause (see function HANGUP_CAUSE).
I want
2013/3/25 Olivier oza_4...@yahoo.fr
2013/3/25 Matthew Jordan mjor...@digium.com
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such as
Asterish and Tech
Hi,
I've got a brand new Asterisk 11 setup for which I would like to keep the
number of loaded modules to a minimum.
My goal is to this setup in a pure SIP environment, for switching incoming
calls to outgoing tSIP trunks.
When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an
Hi,
In my lab, I'm testing BRI spans in NT/ptmp mode.
My setup is:
asterisk 11.2.1
libpri 1.4.14
dahdi 2.6.1
wctdm24xxp (HA8 hybrid with B400M)
SIP phone Asterisk with HA8 Patton SN4638 Asterisk
SIP phone
The single BRI line I'm testing remains down:
CLI pri show spans
PRI
hello,
In a machine I've got :
CLI pri set debug off
No such command 'pri set debug off' (type 'core show help pri set' for
other possible commands)
CLI core show help pri
pri intense debug span no description available
pri service disable channel Remove a channel from service
pri
2013/3/3 Mc GRATH Ricardo mcgra...@mail2web.com
Hi Olivier
It seems wrong configuration, because according to your mail Asterisk it
will be acting as terminal mode (ie Patton gateway acting as network and
asterisk as terminal).
But Asterisk message is indicated Asterisk a s a NT mode
Hello,
On a production system, I'm seeing this:
[Feb 13 16:47:00] WARNING[14742] res_agi.c: Unable to create toast pipe:
Too many open files
[Feb 13 16:47:00] WARNING[9283] acl.c: Cannot create socket
[Feb 13 16:47:00] WARNING[9283] rtp.c: Unable to allocate RTCP socket: Too
many open files
[Feb
2013/2/15 Olivier oza_4...@yahoo.fr
Hello,
On a production system, I'm seeing this:
[Feb 13 16:47:00] WARNING[14742] res_agi.c: Unable to create toast pipe:
Too many open files
[Feb 13 16:47:00] WARNING[9283] acl.c: Cannot create socket
[Feb 13 16:47:00] WARNING[9283] rtp.c: Unable
Hi,
Reading comment in the bottom of
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk,
I thought I could install asterisk 11 in non-standard locations such as
/usr/local/sbin simply typing (from source directory):
make install DESTDIR=/usr/local/sbin
Doing so seems to
, it looks like but it doesn't behave as it looks ;-)
** **
Jacob
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Tuesday, February 12, 2013 6:03 AM
*To:* Asterisk Users Mailing List
2013/2/12 Doug Lytle supp...@drdos.info
non-standard locations such as /usr/local/sbin
If compiling from source, it'd normally be specified by the --prefix
option:
./configure --prefix=/usr/local
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
2013/2/1 Don Kelly d...@donkelly.biz
-Original Message-
snip
What I had in mind is to use someone's cellphone as a presence detector.
Let me explain:
- as the first thing you take along when leaving a room or location,
is your own cellphone, why not use chan_mobile and a
2013/1/31 Sebastian Arcus s...@open-t.co.uk
On 31/01/13 07:25, Olivier wrote:
Hello,
On a LAN, is it possible to install a bluetooth dongle on one
workstation (at this time, this workstation OS is not specified) and use
it with chan_mobile ?
I've read some USB over IP (or Ethernet
Hello,
On a LAN, is it possible to install a bluetooth dongle on one workstation
(at this time, this workstation OS is not specified) and use it with
chan_mobile ?
I've read some USB over IP (or Ethernet) middleware exist but I'm not
certain I'm looking at the right direction.
Regards
--
-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Friday, January 18, 2013 3:35 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to give users the capability to set
CDR userfield
Hi,
Let say that in a call center, callers are recognized and categorized in 4
priority levels (priority 1 for Very Very Important Personalities, 2 for
VIP, and so on) before entering a Queue.
How can you make sure a priority 2 caller is answered before priority 3
callers, for instance ?
I can
2013/1/25 Alec Davis siva...@paradise.net.nz
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is
2013/1/25 Michael Keuter li...@mksolutions.info
Am 25.01.2013 um 17:22 schrieb Olivier:
Hi,
Let say that in a call center, callers are recognized and categorized in
4 priority levels (priority 1 for Very Very Important Personalities, 2 for
VIP, and so on) before entering a Queue
Hello,
I've seen some desktop SIP phones (Snom, Yealink) intregrate a VPN (OpenVPN
?) client.
Has someone experience to share about that particular feature ?
Is this experience rather successful ?
My underlying question is can one supervise and configure these desktop
phones, in teleworking
Hello,
I've read a couple of docs on the web about distributed call centers
(mainly this one
application map.
I've found old references explaining that this is not supported but I don't
if it's still valid or not.
Any ex
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From:Olivier oza_4...@yahoo.fr
To:Asterisk Users Mailing List - Non-Commercial
Hi,
Let say we have a call center from which agents get calls from both
on-queues and off-queues calls (ie calls passing through queues or
direct calls non passing through queues).
Regulation here specify prior consent before recording call.
How can I best enforce this compliance ?
What would
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set
Hi,
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy offices, I'm wondering if wireless headets
with bluetooth connectivity exist ?
I'm refering to bluetooth as
2013/1/16 Administrator TOOTAI ad...@tootai.net
Le 16/01/2013 12:29, Olivier a écrit :
Hi,
Hello
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy offices
2013/1/16 Administrator TOOTAI ad...@tootai.net
Le 16/01/2013 12:29, Olivier a écrit :
Hi,
Hello
I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially noisy offices
Hello,
I've seen this Atom-based NAS appliance with one PCIe 1x slot :
http://www.thecus.com/product.php?PROD_ID=65
Has someone successfully added a Digium, Sangoma or other telephony card in
it along asterisk of course ?
Regards
--
Hi,
I would like to edit reports showing how fast operator and users answer
incoming calls.
Users are spread over 6 locations, each with its own asterisk instance.
Operator is on main site.
Users have casual extension but operator logs as queue agent.
I've read or/and tried Star2Billing's
Hi,
Have you experienced Asterisk 11 in production ?
What do you think of it ?
Which libpri version, if any, did you then associate with Asterisk 11 ?
Regards
--
_
-- Bandwidth and Colocation Provided by
Hello,
I need to develop an application that will query (mostly reading) an
existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx 2.10 install
on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.
The tools I'm planning
reason, a
uniqueid is missing, it would still work ok.
On the other columns I need to search I have created just an index.
Leandro
2012/12/6 Olivier oza_4...@yahoo.fr
Hello,
I need to develop an application that will query (mostly reading) an
existing MySQL CDR database.
This database
Hello,
For an operator, I'm looking for a software application with which operator
would be both able:
- to see the list of awaiting calls,
- to fill a (customizable) form with the name, number and reasonto use
whern returning the call.
Suggestions ?
Regards
--
2012/11/27 Richard Mudgett rmudg...@digium.com
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote:
.. I get errors while trying to compile Libpri 1.4.13. (check
attachment} Can you guys please help me prescribe a fix.
[snip]
gcc -g -Wall -Werror
2012/11/19 Shaun Ruffell sruff...@digium.com
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote:
.. I get errors while trying to compile Libpri 1.4.13. (check
attachment} Can you guys please help me prescribe a fix.
[snip]
gcc -g -Wall -Werror -Wstrict-prototypes
2012/11/15 Frederic Van Espen frederic...@gmail.com
On Thu, 2012-11-15 at 08:52 +0100, Olivier wrote:
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
store this VLAN ID into its flash memory so that, on next boot, it
would broadcast its DHCP request using the VLAN he
2012/11/15 Frederic Van Espen frederic...@gmail.com
On Thu, 2012-11-15 at 08:53 +0100, Hans Witvliet wrote:
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will cause problems
If your extension does not start with an underscore, it is not
considered
Hello,
As I don't have any Polcyom SoundPoint at hand, at the moment, I would like
the following to this list readers.
I have a steup where two DHCP servers are installed : one dedicated for IP
telephony, the other for IT..
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
Hello,
I have the following case.
A customer is a heavy Meetme/audio conference user.
He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference
station).
Users complain they often do not hear the other party loud enough.
The setup is then:
Remote party --- PSTN/ISDN--- Asterisk
2012/11/7 Jeff LaCoursiere j...@sunfone.com
Just to chime in, if you REALLY want multi-tenant, it is super easy and
surprisingly efficient to use kernel level virtualization to run multiple
instances of asterisk (and even FreePBX). We use LXC to do this. The
host runs an instance that has
2012/10/31 Benny Amorsen benny+use...@amorsen.dk
Olivier oza_4...@yahoo.fr writes:
That's the point : to me, casual @pickupmark mechanism don't work with
calls that entered into a queue : the extension rings but you can't pick
the call up with a directed pickup.
(For general pickup
2012/10/30 Benny Amorsen benny+use...@amorsen.dk
Mitch Claborn mitch...@claborn.net writes:
In our sales queue, we have wrapup time set to 15 seconds. When the
phones are really busy, the operators would like the ability to bypass
that 15 second wait and grab the next call in the queue.
up with a directed pickup.
(For general pickup, that's another strory).
(and I would be very pleased to be wrong)
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Tuesday, October 30, 2012 4:22 PM
2012/10/11 Kinsey Moore kmo...@digium.com
On 10/11/2012 10:31 AM, Olivier wrote:
2012/10/11 Kinsey Moore kmo...@digium.com mailto:kmo...@digium.com
Hi Olivier,
My questions are:
1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but
I can
upgrade to 1.8.17
Hi,
Watching my queue_log file content, I can read entries like :
1349940957|1349940957.4|6025|Local/6455@from-queue/n|ADDMEMBER|
My extensions.conf file used this statement:
exten = *9876,n,AddQueueMember(6025,Local/6455@from-queue
/n,10,,FOOBAR,hint:6455@ext-local)
I was expecting to see:
2012/10/11 Kinsey Moore kmo...@digium.com
Hi Olivier,
My questions are:
1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but I can
upgrade to 1.8.17 if, and only if, necessary) so that ADDMEMBER entries in
queue_log refers to member name instead of member location ?
If positive
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id=152230atid=783657
Has anyone been successful when integrating latest version of Asterisk (10
or 1.8,
2012/10/8 Michel Verbraak mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id
2012/10/8 Michel Verbraak mic...@verbraak.org
Op 08-10-12 09:24, Olivier schreef:
Hi,
I've read this thread in this list history
http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/261151/match=t38modem
http://sourceforge.net/tracker/?func=detailaid=3337581group_id
Hi,
1. I've got this question bouncing in my mind for a long time: why are
alarm transmitters often said to be avoided with DSL lines ?
The kind of alarm transmitters I'm thinking about are those having two
analog ports: one connected to Telco analog line, the other to a fax or a
terminal or
2012/10/4 Brett Lehrer brett.leh...@solarismed.com
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming
and outgoing faxes, I'm getting a failure rate of just over 25%, and over a
handful of
2012/10/4 Brett Lehrer brett.leh...@solarismed.com
What is the setup you're talking about ?
Is it something like this ?
PSTN nexVortex T.38 gateway - Internet - DSL modem ---
Asterisk Fax machine
Olivier,
Sorry, I did a poor job explaining that. That's basically correct
2012/10/1 Mark Michelson mmichel...@digium.com
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/**jira/browse/ASTERISK-20163https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, September 13, 2012 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
Hello,
I want to offer SIP phone user a custom fax-to-email feature.
Here is how I would describe this feature:
- for every SIP phone,a custom email address is defined
- when a SIP phone answers an incoming call (from a trunk or another SIP
endpoint), it detects the call is coming from a fax
2012/9/13 Shaun Ruffell sruff...@digium.com
On Wed, Sep 12, 2012 at 11:52:40PM -0400, Mark Robinson wrote:
I know that asterisk on virtual machine require a timing source.
What would you suggest to use for timing? We will plan to use only
SIP and IAX2.
If you're on a newish kernel
2012/9/13 bilal ghayyad bilmar...@yahoo.com
Hi All;
Is there a module (addon or already built in) that enable us to receive
the fax on the telephony card and save it as image (or any other format)
and sent it to email?
have a look at receivefax (core show application ReceiveFAX)
2012/8/31 Jeff LaCoursiere j...@sunfone.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, August 28, 2012 3:24 PM
To: asterisk-users@lists.digium.com
Subject:
2012/8/21 isr...@gmail.com
She's talking about asterisk 11 not asterisk 1.8.11
-Original Message-
From: Phil Frost p...@macprofessionals.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 21 Aug 2012 15:19:31
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users
2012/9/13 Benedikt Schöffmann benedikt.schoeffm...@gmail.com
Hi there,
I'm setting up a Asterisk network and I ran into some problems ... as you
might have guessed :)
The set up is like this:
Internal Communication in the company should be handled through softphones
over an asterisk
2012/8/16 Eric Wieling ewiel...@nyigc.com
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static
basis,
For curiosity's sake, could you make it work first using static settings ?
but no way I've been
Hello,
In Freepbx 2.10, for instance, you can read several instances of the
following pattern :
In extensions.conf
#include extensions_additional.conf
#include extensions_custom.conf
In extensions_additional.conf
[macro-hangupcall]
include = macro-hangupcall-custom
exten = s,1,Noop(foobar1)
Hi,
I'm banging my head on Freepbx 2.10 setup with which a SIP hardphone can
subscribe to some Freepbx-generated hints and still cannot subscribe to
other Freepbx-generated hints (404 Not Found).
I would be very curious to learn here a bit more about how Asterisk 1.8
(and above) deals with hint
PS: Another question
Let my system is configured with 2 hints like this :
*2711@timeconditions-toggle: Custom:TC11
State:InUse Watchers 0
6452@ext-local :
SIP/6452 State:Unavailable Watchers 0
Let say I
2012/8/19 Alec Davis siva...@paradise.net.nz
Do you also know why it hasn't been accepted ? Seems like this
functionality is asked for on different forums. Wanting
to watch a
queue for calls is not that strange.
Not sure why?
Maybe I didn't promote it enough.
Hi,
I'm benchmarking the performance of a Patton Smartnode 411X gateway.
My setup is :
GSM phone --PSTN-- SN411X --SIP-- Asterisk --SIP-- SIP phone
My reference setup is:
GSM phone --PSTN-- analog phone
In the first case, it takes roughly 10s from the moment GSM user hits Send
button to the
2012/7/5, giovanni.v i...@keybits.org:
The matter becomes more difficult approaching a building install as
there are no devices to handle properly that.
I think the snom PA-1 may be a good candidate to play with because of a
versatile I/O that could be interfaced to a custom door-phone bridge
2012/7/5, C F shma...@gmail.com:
snip
no sure if you can have it function
such that any number entered will actually be send to a gateway.
To me, that is the key selling point :
people are used to just dial a number and then press a Fax button that
I can't succeed in anything more than
2012/7/5, C F shma...@gmail.com:
I searched a bit more,
http://www.muratec.com/catalog/F320_config.html#email
The above model supports t.37
That's very interesting to know.
I quickly googled for t.37 and found several other vendors mentioning
this (some from rather old documents).
The strange
Hi,
Now and then, I'm facing environments in which it could be helpful to
integrate building intercom systems with Asterisk.
Those intercom systems are made of :
- a main panel, showing company names and equiped with a speaker, a
microphone and an optional video cam
- a doorstrike
- several
Hi,
I'm curious about the availability of Multi Function Printers with the
following feature :
- user feeds paper sheets in
- user dials a phone number (0123456, for instance) then a hits single button
- the result is that the paper sheets are scanned into a file which is
emailed to a given
2012/6/26, Richard Mudgett rmudg...@digium.com:
This is the option I will try.
I'll report my findings here.
My findings, after setting layer1_presence=ignore in chan_dahdi.conf
are :
== Starting D-Channel on span 1
== Starting D-Channel on span 2
== Primary D-Channel on span 1 up
2012/6/22, Olivier oza_4...@yahoo.fr:
2012/6/21, Richard Mudgett rmudg...@digium.com:
My previous message was incomplete.
On thing to note is I had to forbid hfcmulti in modprobe.d in the
second box to comply with a warning from dahdi. Without that, I could
see this line in the output
As basically Asterisk 1.8 requires libpri1.4.11 and up, I had to
downgrade Asterisk to 1.6 and libpri to 1.4.10.2 to get pri show
spans working back again.
Having a 1.4.13 published with complete PtmP support will be much appreciated.
Regards
--
2012/6/25, Tzafrir Cohen tzafrir.co...@xorcom.com:
On Fri, Jun 22, 2012 at 08:07:54PM +1200, Alec Davis wrote:
Have a look at the latest blacklist sample in dahdi trunk
http://svnview.digium.com/svn/dahdi/tools/trunk/blacklist.sample?view=log
file: blacklist.sample
...
# Some mISDN drivers
. It is also the reason for the
layer1_presence and layer2_persistence options.
4. Waht would you suggest ?
Regards
2012/6/21, Olivier oza_4...@yahoo.fr:
Hi,
After an upgrade, I discovered yesterday strange things I would
like
to share here.
Basically, I'me comparing platforms
physically removing BRI card.
I was a bit luckier thanks to default dahdi.blacklist.conf file.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, 21 June 2012 9:37 p.m
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