From the dnsmasq Manpage:
-O,
--dhcp-option=[network-id,[network-id,]][encap:opt,][vendor:[vendor-class],][opt|option:opt-name],[value[,value]]
Specify different or extra options to DHCP clients. By default, dnsmasq sends
some standard options to DHCP clients, the netmask and broadcast address
JR Richardson wrote:
On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
On Monday 28 December 2009 18:09:15 JR Richardson wrote:
I turned on console debug to see the actual mysql queries and to my
surprise and concern, I see every query for an extension priority
repeated
On Tue, Dec 29, 2009 at 11:06:40PM -0700, Mike Diehl wrote:
Hi all,
I'm trying to use a wrt54gl router running dd-wrt as a provisioning server
for
a remote installation.
I've got dhcp working and I have provisioning files ready to go. I
understand
that I need to set bootp option 66
Taylor, Jonn skrev:
Leif Neland wrote:
I can't believe anyone would use RJ-11 any more. You can multi-purpose
RJ-45 jacks to work with POTS lines. Run everything down to a central
panel and send pots over the jacks that you need to. That way if you
decide you need/want to go IP in the
Tim Nelson skrev:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using the BT201 phones from Grandstream for this purpose.
In fact, this is the only
Hello!
I run asterisk 1.6.1.0, dahdi 2.1.0.4 with TE122.
I always (and only) have missed interrupt when dahdi disables echo
canceller (ng2 or oslec- no difference).
Dec 29 14:00:54 asterisk kernel: dahdi: Disabled echo canceller because
of tone (rx) on channel 1
Dec 29 14:00:54 asterisk
Hello,
I have enabled call parking and it works great.
However, when the hold time hits the parkingtime, the extension that
parked the call is called back.
The problem is, if that extension does not pickup the returning call, the
call gets dropped.
Is it possible to get Asterisk to
Dear All
I want to enable festival text-to-speech . To this end , I added the
required lines to festival.scm but when I want to start festival server I
face with the following error :
#festival --server
SIOD ERROR: end of file inside list
Closing a file left open: /usr/share/festival/festival.scm
hi all,
I have a little problem I'm using asterisk with opensips as opensips
dispatches calls to asterisk. I have to use multiple asterisk servers but
since for the time being im using 1 machine for testing i want to run
different instances of asterisk running on 1 pc listening to different
On Wed, Dec 30, 2009 at 5:29 PM, Saeed Akhtar saeedakhtar@gmail.comwrote:
hi all,
I have a little problem I'm using asterisk with opensips as opensips
dispatches calls to asterisk. I have to use multiple asterisk servers but
since for the time being im using 1 machine for testing i
Dan Journo wrote:
Hello,
I have enabled call parking and it works great.
However, when the “hold time” hits the “parkingtime”, the extension
that parked the call is called back.
The problem is, if that extension does not pickup the returning call,
the call gets dropped.
Is it
Hi Doug,
Where did you paste that from?
I'm using 1.4.
Regards,
Dan
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Dear All
Can you please give me more hint on how Asterisk Dictate() works?
Thank you
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Dan Journo wrote:
Hi Doug,
Where did you paste that from?
I'm using 1.4.
From my dial plan, I'm also using 1.4.x
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I am using asterisk 1.6.0 and -- not all the time -- when a caller comes
in and my ivrdials an extension, the ring he gets sounds like a modem
handshake instead of the normal ring tone and it only sounds once even
if the phone is not picked up. Anyone seeing this -- the logs look fine
as far as I
Google is your friend. You should use it. Search for:
asterisk extensions.conf dictate
or
asterisk extensions.conf dictate example
Some results:
http://www.asteriskguru.com/tutorials/dictate.html
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dictate
On Wed, Dec 30, 2009 at
Or you could setup a VPS environment (perhaps openvz) and run Asterisk
in a virtual environment.
I've done this in the past and it works well.
ram wrote:
On Wed, Dec 30, 2009 at 5:29 PM, Saeed Akhtar
saeedakhtar@gmail.com mailto:saeedakhtar@gmail.com wrote:
hi all,
I
On Wed, Dec 30, 2009 at 04:59:11PM +0500, Saeed Akhtar wrote:
hi all,
I have a little problem I'm using asterisk with opensips as opensips
dispatches calls to asterisk. I have to use multiple asterisk servers but
since for the time being im using 1 machine for testing i want to run
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve this:
voicemail.conf
4000 = 4000,system,voicem...@net
Relevant extensions.conf line:
exten =
Hi,
I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card.
Everything is working fine except the caller ID of incoming call from PSTN
line. The phone display is showing Unknown when there is an incoming call.
*My log file showing this while an incoming call on PSTN line:*
tail -f
listu...@spamomania.co.uk wrote:
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve this:
And this custom .gsm file is a greeting? Or, you're
On Wed, 2009-12-30 at 14:56 +, listu...@spamomania.co.uk wrote:
It all works fine, playing the system VM greating, but I would like to
use the custom .gsm for this user only. Can anyone help?
The greetings and voicemail messages are typically stored in
the /var/spool/asterisk/voicemail
This might work:
Change
exten = 2,n,VoiceMail(4...@voicemail)
to
exten = 2,n,playback(custom)
exten = 2,n,VoiceMail(4...@voicemail,s)
The s flag makes voicemail play no greeting instead of the default b
(busy) or u (unavail)
-Original Message-
From:
How is DAHDI-1 set up in users.conf?
You need something like this
; Span 2: WCTDM/4 Wildcard TDM400P REV I Board 5
[4001]
fullname = Line 1
cid_number = 5551212
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun
I have an Asterisk 1.4.2 server with 3 different SIP providers and
Asterisk for Skype gateway installed. Periodically the SIP providers go
offline for some reason, or the Skype connection fails.
When this happens, I lose my SIP registration to the provider.
Unfortunately I don't know this has
You need to wait at least 1 second on an incoming POTS line for CID info, add a
wait(1) as the first step on incoming connections.
Thank you and have a nice day,
Anthony Francis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun
On Tue, 3 Feb 2009, C. Savinovich wrote:
Asterisk compatible One to many video is achieved with VidPhone. You can
download the web embedded video component free by signing up an account on
my website www.itntelecom.com. Any help on usage, just send me a note and I
will be glad to help you
Get the customer to log into their voicemail mailbox and follow the
instructions to record an unavailable message (Options 0 then 1 if there
are no messages I think)
Then in the conf you need
exten = 2,n,VoiceMail(4...@voicemail,u)
Ish
listu...@spamomania.co.uk wrote:
I'm struggle to
On Wed, Dec 30, 2009 at 09:05:13AM -0600, Danny Nicholas wrote:
How is DAHDI-1 set up in users.conf?
users.conf? with freepbx?
FXS extensions in FreePBX sadly need to be set up in the GUI (no usable
command-line interface). If you just add them (indirectly) through
chan_dahdi.conf, you'll miss
On Wed, 30 Dec 2009, Leif Neland wrote:
Tim Nelson skrev:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using the BT201 phones from Grandstream for this
purpose.
On Wed, 2009-12-30 at 10:00 -0500, Doug Lytle wrote:
listu...@spamomania.co.uk wrote:
I'm struggle to answer a simple question. One user at extension 4000
wants a custom .gsm file to play for their mailbox. I can't figure where
to put it/what to set in voicemail.conf to achieve this:
On Wed, 2009-12-30 at 15:18 +, Ishfaq Malik wrote:
Get the customer to log into their voicemail mailbox and follow the
instructions to record an unavailable message (Options 0 then 1 if there
are no messages I think)
Then in the conf you need
exten = 2,n,VoiceMail(4...@voicemail,u)
listu...@spamomania.co.uk wrote:
On Wed, 2009-12-30 at 15:18 +, Ishfaq Malik wrote:
Get the customer to log into their voicemail mailbox and follow the
instructions to record an unavailable message (Options 0 then 1 if there
are no messages I think)
Then in the conf you need
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
database structure for cdr_mysql is:
CREATE TABLE cdr (
calldate datetime NOT NULL default '-00-00 00:00:00',
clid varchar(80) NOT NULL default '',
src varchar(80) NOT NULL default '',
dst varchar(80) NOT NULL default
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
database structure for cdr_mysql is:
CREATE TABLE cdr (
calldate datetime NOT NULL default '-00-00 00:00:00',
clid varchar(80) NOT NULL default '',
src
Thank you Mr.Antony Francis for the reply. Actually where to add that
wait(1) in the server?. Please reply in detail about this.
Regards,
Aruns
On Wed, Dec 30, 2009 at 8:48 PM, Anthony Francis - Handy Networks LLC
anth...@handynetworks.com wrote:
You need to wait at least 1 second on an
At 07:05 AM 12/30/2009, you wrote:
I am using
asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The
phone display is showing Unknown when there is an incoming
call.
My log file showing this while an incoming
Tilghman Lesher wrote:
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
database structure for cdr_mysql is:
CREATE TABLE cdr (
calldate datetime NOT NULL default '-00-00 00:00:00',
clid varchar(80)
Wednesday, December 30, 2009, 6:48:37 PM, Robert wrote:
Tilghman Lesher wrote:
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
Just curious if anyone has successfully patched cdr_addon_mysql to use
accept the latest cdr fields from 1.4 ... namely: 'start', 'answer', 'end'?
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
Hi All,
Can any one please help me with the Dial plan for the case I explained below...
From: Aditya Kumar adityakumar...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Mon, December 28, 2009 10:10:23 AM
Subject: Asterik with out registration.
Hi,
Can I
On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland
thermalwetl...@gmail.com wrote:
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP
On Wednesday 30 December 2009 11:48:37 Robert Broyles wrote:
So my next question is could I take the cdr_mysql from 1.6's addons and
use it in 1.4?
No. The APIs are significantly different enough that a backport would
require a good amount of modification. However, there is a backport of
On Wed, Dec 30, 2009 at 8:27 AM, Matt Darnell mattdarn...@gmail.com wrote:
# Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer
on the receiving side of a SIP channel. (Added in Version 1.4)
# Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on
the receive
At 12:36 PM on 30 Dec 2009, hadi motamedi wrote:
Dear All
Can you please give me more hint on how Asterisk Dictate() works?
Thank you
http://lmgtfy.com/?q=asterisk+dictate
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0
Gordon Henderson skrev:
On Wed, 30 Dec 2009, Leif Neland wrote:
Tim Nelson skrev:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using
I would like to display the status of my sip trunk channel, so by the
led of the BLF button I can see the line is busy.
(I'm using a Patton ISDN gateway).
Someone can explain me how to configure dialplane to hint the sip trunk?
Thanks in advance
___
Exten = 123,hint,SIP/X
Where 123 is the hint number to watch and X is the number of your SIP
line/extension.
This goes in the [internal] context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio
Sent:
You are correct.. I had the correct context on my current production
configuration I just copied from an older saved file.. So the
[enterbroadvoice] has a context of incoming and incoming is defined in the
extensions.con. But still have the same problem with incoming jumping
directly into
http://www.techcrunch.com/2009/12/30/twilio-raises-3-7-million-for-power
ful-telephony-api/
wow really?
Cheers,
Dean
___
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asterisk-users mailing list
To
On Wednesday 30 December 2009 19:19:02 Dean Collins wrote:
http://www.techcrunch.com/2009/12/30/twilio-raises-3-7-million-for-power
ful-telephony-api/
wow really?
Dean-
Do you think you could keep this on the -biz list, please, or just Twitter it
to your followers? This has nothing to do
Dean Collins wrote:
http://www.techcrunch.com/2009/12/30/twilio-raises-3-7-million-for-powerful-telephony-api/
wow really?
Love the Ericsson PTT rotary phone.
Guess they are focusing on the Euro market?
John Novack
Cheers,
Dean
At 18:22 12/30/2009, Qurba Joog wrote:
You are correct.. I had the correct context on my current production
configuration I just copied from an older saved file.. So the
[enterbroadvoice] has a context of incoming and incoming is defined
in the extensions.con. But still have the same problem
Hi,
Thanks for the reply. Actually I am using Astrisknow 1.5. So I added *exten
= s,1,wait(1)* in extensions_custom.conf file.It is now looks like this
*[from-internal-custom]
exten = 1234,1,Playback(demo-congrats) ; extensions can dial 1234
exten = 1234,2,Hangup()
exten =
Dear All
Can you please let me know if we can have different codec schemes for
audio codec in audio codec out ? I mean , in one application , we
can have our audio codec input set to G.711 a-law and our audio codec
output set to G.711 u-law . I am facing with an application that calls
for such a
On Wed, Dec 30, 2009 at 7:11 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Dec 30, 2009 at 04:59:11PM +0500, Saeed Akhtar wrote:
hi all,
I have a little problem I'm using asterisk with opensips as opensips
dispatches calls to asterisk. I have to use multiple asterisk
After using the CentOS repo's at digium to install dahdi Linux tools, I got
this:
Installing : kmod-dahdi-linux-fwload-vpmadt032
WARNING: /lib/modules/2.6.18-164.9.1.el5/dahdi/dahdi_vpmadt032_loader.ko needs
unknown symbol voicebus_transmit
WARNING:
Since you have 'answered' the caller's call, so the ring cannot be sent as a
signal to the caller.. Instead, it is sent in media. Asterisk does the ring
generation in audio for you.
You can control and configure the type of ring that asterisk generates by
configuring indications.conf
To start
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