On Mon, 2011-01-24 at 01:09 -0500, RR wrote:
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger
pabelan...@digium.com wrote:
On 11-01-23 10:24 PM, RR wrote:
email from Kevin Flemming talking about =2.6.27 so thought
I'd ask esp. coz
I have 2.6.26-2 yet I don't
On Mon, Jan 24, 2011 at 3:11 AM, Stelios Koroneos
skoron...@digital-opsis.com wrote:
On Mon, 2011-01-24 at 01:09 -0500, RR wrote:
On Mon, Jan 24, 2011 at 12:24 AM, Paul Belanger
pabelan...@digium.com wrote:
On 11-01-23 10:24 PM, RR wrote:
email from Kevin Flemming
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to squeeze.
R
--
On Saturday 22 Jan 2011, Tim Panton wrote:
I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card.
Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable?
If so, any clues where I might buy one in the UK? The Digium card sellers
don't seem to stock such
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West ro...@firedrake.orgwrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock
lenny
installation with the updated glibc as well as the latest kernel
At this point the
On Monday 24 January 2011 03:46:18 A J Stiles wrote:
On Saturday 22 Jan 2011, Tim Panton wrote:
I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1
card.
Am I right in thinking that I'll need a special 'crossover-E1' RJ45
cable?
If so, any clues where I might buy
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so
On Monday 24 January 2011 04:09:31 Olivier wrote:
2011/1/23 Tzafrir Cohen tzafrir.co...@xorcom.com
On Thu, Jan 20, 2011 at 06:22:23PM +, A J Stiles wrote:
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script
On 11-01-23 02:56 PM, Jeff B wrote:
There does not seem to be very much info out there about using LDAP to
create asterisk configurations. Does anyone have some information
that they would suggest I start with?
We've tried to document some of it here:
On Monday 24 Jan 2011, Tilghman Lesher wrote:
On Monday 24 January 2011 03:46:18 A J Stiles wrote:
white/blue blue white/green orange white/orange green white/brown brown
This is incorrect. The pairs should be:
blue white/blue white/green white/orange orange green white/brown brown
Wire
Am 23.01.2011 18:38, schrieb Carlos Chavez:
On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez
cur...@telecomabmex.com wrote:
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote
On 22 Jan 2011, at 18:02, Carlos Chavez wrote:
Cannot
Thanks, That's more info than I've been able to find to date. I'll
work on digesting it now.
On Mon, Jan 24, 2011 at 7:37 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-01-23 02:56 PM, Jeff B wrote:
There does not seem to be very much info out there about using LDAP to
create
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.orgwrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock
lenny
installation with the
On Mon, Jan 24, 2011 at 10:22 AM, Jeff B jeffb.l...@gmail.com wrote:
Thanks, That's more info than I've been able to find to date. I'll
work on digesting it now.
Please add you comments and findings to
https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver to
help us have a good
Hello list,
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
I have a 'cdr' table in my MySQL-DB. On this table the user
'asteriskcdr' has select, insert,
Hello.
I am running Asterisk v1.6 on Ubuntu 10.10 with FreePBX v2.8.
When I call the voicemail for any of my extensions, the call just dies. On a
softphone, I get no sound whatsoever; it just hangs up after a couple of
seconds. On my handset attached to my SPA-3102, it get a sound like when
Hi,
maybe the error is on this line:
sock=/tmp/mysql.sock
if you use CentOS the correct line is:
sock=/var/lib/mysql/mysql.sock
if you use Debian/ubuntu:
sock=/var/run/mysqld/mysqld.sock
Regards--
_
-- Bandwidth and
Jonas Kellens wrote:
Hello list,
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
I have a 'cdr' table in my MySQL-DB. On this table the user
On 01/24/2011 07:43 AM, Jonas Kellens wrote:
Hello list,
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
I have a 'cdr' table in my MySQL-DB. On this
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org mailto:ro...@firedrake.org wrote:
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com wrote:
On Mon, Jan 24, 2011 at 4:06 AM, Roger Burton West
ro...@firedrake.org
In the meantime, does anyone have a nice way to update a stable/stock lenny
installation with the updated glibc as well as the latest kernel
Scary and risky, as others have noted!
There is an official backports release kit associated with Debian,
which contains newer versions of many packages
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote:
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries.
How would
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Monday, January 24, 2011 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] U-verse DTMF
On Mon, Jan 24, 2011 at 3:37 PM, Steve Edwards
asterisk@sedwards.com wrote:
One of my clients is complaining that their customers that use U-verse
(and other cable providers) for telephone service cannot enter credit
card numbers reliably.
The issue not all digits are received in my
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
We know the problem exists -- the boss just installed U-verse at his house
:)
It works fine from cell and copper, just not from U-verse and their ilk.
Well, I would say more data samples are needed then. It could
On Mon, 24 Jan 2011, David Backeberg wrote:
Well, I would say more data samples are needed then. It could certainly
be the boss's uverse connection and not other uverse connections.
They've had complaints from off-site agents and from long term customers
who noticed that 'it stopped working'
On 01/24/2011 03:54 PM, David Backeberg wrote:
Well, I would say more data samples are needed then. It could
certainly be the boss's uverse connection and not other uverse
connections.
Have you ruled out a problem on the U-Verse side? In particular, do
U-Verse customers only experience
I'm confused about a few things relating to realtime, SIP and config in
general.
As I understand it, with the exception of extensions.conf, I can either
have a config file completely in text or completely in a database. Is
that correct? I can't find documentation for exactly what switch = does
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at 4:56 AM, RR ranjt...@gmail.com
mailto:ranjt...@gmail.com
Does anyone know how to get rid of these warnings?
~
[Jan 24 18:10:04] WARNING[2629]: chan_sip.c:15898
handle_request_subscribe: SUBSCRIBE failure: no Accept header: pvt:
stateid: -1, laststate: 0, dialogver: 0, subscribecont:
I have continued searching but still haven’t found any way to get asterisk to
send RTCPs when on hold.
This issue seems to have come up several times and has been reported by several
people but nothing seems to have come of it. Should I be filing a bug report?
Or are there any workarounds
Hi List.
Have any of you guys ever see an incoming call throught Dahdi channel which
has an callerid T.
I know whenever is a private call, it shows as callerid, but what does it
mean a T callerid?
Best Regards
--
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
--
On Mon, Jan 24, 2011 at 7:07 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/24/2011 12:46 PM, RR wrote:
On Mon, Jan 24, 2011 at 12:09 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 01/24/2011 07:29 AM, RR wrote:
On Mon, Jan 24, 2011 at
Hi
I want to solve ipaddress with DNS A queries.
There is the problem that ipddress resolution of the DNS is not correct.
Because Ａsterisk is not put sip. before a message of XX.com
I could see in the log that asterisk send A queries without sip.
Is this problem for asterisk configuration or
I know this is an {*} list but does anyone know if simply adding the Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having to rebuild the system from scratch?
In my experience: you're likely to run
On Mon, 24 Jan 2011, David Backeberg wrote:
Can you record a few calls just to confirm the problem?
If I record via mixmonitor(), I just get a bunch of clicks where the DTMF
should be. I'm assuming this is because the DSP has already taken the DTMF
out of the audio stream.
If I record via
On Mon, 24 Jan 2011, David Backeberg wrote:
Can you record a few calls just to confirm the problem?
On Mon, 24 Jan 2011, Steve Edwards wrote:
If I record via mixmonitor(), I just get a bunch of clicks where the DTMF
should be. I'm assuming this is because the DSP has already taken the DTMF
On Mon, Jan 24, 2011 at 8:57 PM, Dave Platt dpl...@radagast.org wrote:
I know this is an {*} list but does anyone know if simply adding the
Squeeze
repository to my sources.lst and running an 'aptitude
upgrade/safe-upgrade/full-upgrade will just upgrade Lenny - Squeeze
without me having
I guess that was the CallerID transmitted by the calling channel.
On Mon, Jan 24, 2011 at 7:31 PM, Jose Flores Galicia floj...@gmail.com wrote:
Hi List.
Have any of you guys ever see an incoming call throught Dahdi channel which
has an callerid T.
I know whenever is a private call, it shows
Hi all,
I'm looking at my options for getting access to ISDN ISUP fields from
DDI numbers, when connecting to a 3rd party Asterisk server. This is for
a custom voicemail solution, and at this stage I want to avoid renting a
PRI.
The information I need to capture is:
- Calling Number
- Called
Hi All,
i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be
some of you are doing setup and integration on cloud.
below is my setup details which may help you to suggest me solution.
Asterisk version : 1.6.2.6
1) Kamailio server having public_ip as well local ip .i am
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