Hello,
By default there are not much applications on asterisk..
I found some scipts written for tripbox and trying to apply them to
asterisk 1.6 now.
The following code for ex is for DND application.
Here is the entries in my extensions.conf file
This is the application:
[app-dnd-on]
exten =
On Mon, 4 May 2009, David @ULC wrote:
Check this... http://prodsurvey.webng.com/top.jpg
David,
I think the real problem here is that you know very little about Linux
servers and their care and feeding.
The questions you've been asking recently demonstrate that.
This really isn't the place
On Mon, 4 May 2009, Steven J. Douglas wrote:
Maybe you can try leaving out bindport and bindaddr parameters first.
The port defaults to 4569 anyway. As for the bindaddr, you should be
using the IP Address of your interfaces. I am assuming you are using the
IP Address obtained from your
Thanks for the feedback !
I know the IP-address of my Asterisk-server.
The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1).
I have port 4569 forwarded on my NAT/firewall.
Strangely I have the same 'notice' when being attached directly to the internet
(so no firewall in between).
On Sun, 26 Apr 2009 12:51:01 +0100, Tim Panton t...@westhawk.co.uk
wrote:
I'm running asterisk 1.4 on an NSLU2 , only a couple of channels
and minimal transcoding, but it seems fine and stable. £80 + usb storage
Thanks guys for the tips on EdgePBX and the Linksys.
Is the NSLU2 still sold,
Is your box on a public ip or via nat? If eth0 isn't the ip you set it
to bind on it will ignore it.
I mean, is your * box on an internal address?
On 02/05/2009, jonas kellens jonas.kell...@telenet.be wrote:
I have connected my Asterisk-box directly to my internetconnection. I
have disabled my
--[ UxBoD ]-- wrote:
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere now ! I am now getting the following :-
==
On Mon, 4 May 2009, Vincent wrote:
On Sun, 26 Apr 2009 12:51:01 +0100, Tim Panton t...@westhawk.co.uk
wrote:
I'm running asterisk 1.4 on an NSLU2 , only a couple of channels
and minimal transcoding, but it seems fine and stable. £80 + usb storage
Thanks guys for the tips on EdgePBX and
Gavin,
My Asterisk-server has 2 interfaces :
- eth0 = LAN-interface (for SIP-phones to register)
- eth1 = WAN-interface (for IAX-trunking to IAX-provider)
Asterisk is behind NAT (has internal IP-address 192.168.3.248 for WAN_if)
SETUP :
m0n0wall 192.168.3.250 -- 192.168.3.248
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
Thanks,
___
-- Bandwidth and Colocation
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere now ! I am now getting the following :-
== Starting post polarity CID detection
On Mon, 4 May 2009, --[ UxBoD ]-- wrote:
Your BT connector should have only 2 pins. If there are 4 pins on the BT
connector, then it is a modem cable, which is wrong.
Regards,
Steve.
Steve,
You rock! :) Found a BT old cordless phone in the cupboard with a long
cable. Plugged in and
Hello,
In Asterisk 1.4.21, I would like to limit notify of the type of calls.
Explain :
A operator with a panel of BLF for each extension.
An extenions calls any phone through Page() functions.
I would like asterisk does not notify for some calls.
Is this possible ?
--
Antoine
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
The short answer is OrderlyStats isn't really free
On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote:
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better
Hi,
Although not “fanless” appliance, (what is your concern about a fan?) you
could take a look at PIKA Warp, which runs Asterisk and FreePBX, Digium GUI
or any other you are interested in. It supports up to 32 simultaneous
calls.
http://pikawarp.org/ or
In theory, yes. This would be a bounty request for a pbx.c modification.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antoine Patte
Sent: Monday, May 04, 2009 7:09 AM
To: asterisk-users@lists.digium.com
Technoco vdex40 is probably outside of your pricepoint but you might
want to consider them
(and no I don't work for them anymoredoes anyone?)
Lol if someone in NY wants my demo unit
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net
Hi.
Does asterisk support muting per a specific channel?
(like the soft hangup command, were you specify a channel and then
asterisks hangs it up).
1-If it does not, how will one go about to do something like this?
2-how to let the user hear 183 the early media like voice mail prompt
Hello everyone,
I am trying to find out whether AOC (any of the 3 types) is currently
supported by Asterisk and for which hardware combination.
Does anyone know if AOC info can be used in the CDR when the interface to
the outside world is ISDN BRI (not pri) and the internal counterpart
On Mon, 4 May 2009, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dean Collins wrote:
Technoco vdex40 is probably outside of your pricepoint but you might
want to consider them
My experience with the vxex40 was not great. This was about six months
ago. I'd not
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dean Collins wrote:
Technoco vdex40 is probably outside of your pricepoint but you might
want to consider them
My experience with the vxex40 was not great. This was about six months
ago. I'd not recommend one unless I could ensure that I got a
I have a need for an ATA that will register over wifi. *NOT* a DECT phone
or other cordless type phone plugged into a wired ATA.
Not seeing one right off, and following the recent discussions about
compact fanless systems, I thought a custom build-your-own might not only
be useful for my
Jeff LaCoursiere wrote:
why not just put something like a wet11(wireless bridge) and pap2t(2x
fxs) in the same box ?
dev time = 0
cost ~100
those are just two of the many products that would work together to do
what you want.
I have a need for an ATA that will register over wifi. *NOT* a
Anyone know if we take a Cisco phone off of a Call Manager system and
flash it for SIP to demo on Asterisk, can we take it back to Cisco and
Call Manager will remember its MAC address and reflash it back to what
it is supposed to be? I would anticipate with Cisco Discovery Protocol
this
On Mon, 4 May 2009, Jon Pounder wrote:
Jeff LaCoursiere wrote:
why not just put something like a wet11(wireless bridge) and pap2t(2x
fxs) in the same box ?
dev time = 0
cost ~100
those are just two of the many products that would work together to do
what you want.
But that wouldn't be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jeff LaCoursiere wrote:
What were the issues?
There were tons of issues involving the generated configurations (such
as not being able to use attended or unattended transfers) but the worst
issue was that the box would not hang up POTS lines. We
Take a look at wrp400 from Linksys/Cisco. It has 2 fxs, 802.11g and 4 switched
ports + wan.
-Original Message-
From: j...@jeff.net
Sent: Mon, 4 May 2009 19:05:25 + (UTC)
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] wireless ATA
On Mon, 4 May 2009, Jon
Jeff LaCoursiere wrote:
On Mon, 4 May 2009, Jon Pounder wrote:
Jeff LaCoursiere wrote:
why not just put something like a wet11(wireless bridge) and pap2t(2x
fxs) in the same box ?
dev time = 0
cost ~100
those are just two of the many products that would work together to do
what you
On Mon, 4 May 2009, Jeff LaCoursiere wrote:
I have a need for an ATA that will register over wifi. *NOT* a DECT
phone or other cordless type phone plugged into a wired ATA.
[snip]
I would like to run DD-WRT on this beast to handle the wifi-client mode,
and I noticed (though have not
On Mon, 4 May 2009, Jimmy Godbout wrote:
Take a look at wrp400 from Linksys/Cisco. It has 2 fxs, 802.11g and 4
switched ports + wan.
Sigh, I suppose this is exactly what I was talking about :) You guys sure
know how to spoil a good project. Hard to compete with Cisco!
I read a handful
I'm just trying to make a real simple Survey via php. Just want it to
play the Question Files, wait for a response, save the response into the
correct variable and then email it all.
I have no issue playing the audio or emailing. But I can't get it to
wait for digits or to properly capture
Yes, you can flash them back and forth as you require.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Shauger
Sent: Monday, May 04, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Just my .02. In my asterisk (1.4.21.2 using Polycom phones), The read in
AGI doesn't work as expected. This is a simple thing to do in a dialplan
and pass to php for the lifting.
Here's what I would do:
exten = s,1(readq1),Read(digitq1,record/q1,16,skip,1,10])
exten =
On Mon, 4 May 2009, James A. Shigley wrote:
I'm just trying to make a real simple Survey via php. Just want it to
play the Question Files, wait for a response, save the response into the
correct variable and then email it all.
If it's real simple, maybe AGI is overkill. Any particular
On 04/05/09 21:17, James A. Shigley wrote:
I’m just trying to make a real simple Survey via php. Just want it to
play the Question Files, wait for a response, save the response into the
correct variable and then email it all.
Packt sent me a book to review recently: Asterisk AGI Programming.
As for queue_log analyzers, you can also look at
http://stats.asternic.org/ . I do not want to give you an opinion
because I wrote it myself. There is a fully functional free version.
Best regards,
--
Nicolás Gudiño
Buenos Aires - Argentina
On Mon, May 4, 2009 at 9:15 AM, Louis-David
David,
Will it happen automatically when you reconnect it to Cisco Call
Manager or does it require additional steps?
Thanks!
On May 4, 2009, at 4:14 PM, David Gibbons wrote:
Yes, you can flash them back and forth as you require.
From: asterisk-users-boun...@lists.digium.com
When the phone is plugged back in to CallManager network, it should
get handed a TFTP server via DHCP, and should automatically download
the configuration from CallManager which includes what firmware to
load. It should then reload the SCCP firmware (if you are not using
SIP) and reboot back to
Jonathan,
Thank you. That is what I assumed but I just wanted to verify it.
On May 4, 2009, at 7:36 PM, Jonathan Thurman wrote:
When the phone is plugged back in to CallManager network, it should
get handed a TFTP server via DHCP, and should automatically download
the configuration from
*resolved*
I had forgotten to load the zaptel drivers prior to the running conf wancfg.
I ended up having to go back to Zaptel, I couldn't get wanpipe to work
with DAHDI.
I don't like DAHDI anyway... even if it is just the name. Gets me
confused with DUNDi and other fail acronyms.
0_0
On Sun,
Hey Gang,
Trying to figure out how I can do the following (have each part working
individually but drawing a blank on combining)
1) click on-screen which sends an AMI originate (works fine)
2) the originated call is to an internal extension that looks up the number
to be dialed (works)
3) then
I'm trying to get asterisk cdr_odbc configured, but it can't connect
through my odbc driver.
switchboard*CLI module load cdr_odbc
[May 4 20:06:04] ERROR[17758]: cdr_odbc.c:358 odbc_load_module:
cdr_odbc: Unable to connect to datasource: asterisk
/etc/odbcinst.ini:
[MySQL]
Description =
Would anyone have a copy of the latest firmware release for the grandstream
BT102 phone? seems grandstream no longer offers it on their website (of if
I missed something a link would be much appreciated.)
Thanks,
Eric
___
-- Bandwidth and Colocation
Receiving a fax with 1.6.1:
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
--- SIP read from 192.168.32.245:5060 ---
SIP/2.0 481 CallLeg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport
From: asterisksip:aster...@192.168.32.16;tag=as2ff08179
To: sip:5...@192.168.32.245:5060;user=phone;tag=c0a80101-2ce1bc03
Call-ID:
On 5/05/2009 1:10 p.m., J. G. wrote:
Hey Gang,
Trying to figure out how I can do the following (have each part working
individually but drawing a blank on combining)
1) click on-screen which sends an AMI originate (works fine)
2) the originated call is to an internal extension that looks up
On Sat, Apr 25, 2009 at 06:03, sean darcy seandar...@gmail.com wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
I have been using Asterisk on 64-bit and 32-bit openSUSE
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