You'd think they'd actually have something like this. But nope, they don't.
Only for debug, but no verbose output filtering.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: December 5, 2008 11:00 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hamachi will be releasing something for Windows Mobile.
They also have something for a Nokia, check out their download link:
https://secure.logmein.com/labs.asp
Also, if you do some Google searching you will see lots of commercial VPN
offerings for Mobile devices.
-Original Message-
queuestats?
Original Message
Subject: Re: [asterisk-users] Fresh installed box
From: "Matt Gibson" [EMAIL PROTECTED]
Date: Fri, October 24, 2008 6:16 pm
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
asterisk-users@lists.digium.com
after a fresh installation
Something very similar had happened to our Polycom's. Somehow a qualify=yes
for all those peers seemed to solve it.
Try it if it's not enabled already.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: October 18, 2008 3:16 PM
To:
Very well put.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: October 5, 2008 1:07 PM
To: Asterisk Users
Subject: Re: [asterisk-users] OT: text/plain
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen
it on the home screen.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 4:03 PM
To: 'Asterisk Users Mailing
into Dial command ?
[Mark Hamilton] snip
why you people need this thing in dial command which can possible with
sip.conf callerid options
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25
This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
phones. Touch Diamond for instance.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 2:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?
This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
I'm game. It's just perfect the way it is - long overdue!
On my behalf, and behalf of the community (hopefully?), thanks a lot Mr.
Bryan for taking the initiative to get this done.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan
Sent:
I don't see why not, Voip-info is very outdated in most respects.
Most of it with bad examples, dating to Asterisk 1.x era.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah Bryan
Sent: September 29, 2008 11:09 AM
To: Asterisk Users Mailing List -
Nope, doesn't seem to work. all I hear is deadair.
Original Message
Subject: Re: [asterisk-users] Streaming MoH on 1.4
From: "Mark Hamilton" [EMAIL PROTECTED]
Date: Sun, September 14, 2008 8:48 pm
To: "Asterisk Users Mailing List - Non-Commercial Discussion&
it down into
chunks, and then cross off the easy chunks first (or the hard ones,
depending on your preference and the priority of the chunks)
later,
PaulH
Mark Hamilton wrote:
Hi,
I’m looking for a GUI like ARI by LittleJohn Consulting (which is not
being maintained actively anymore
. But now I be dreaming.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: September 9, 2008 5:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: ARI
Paul,
Thank you very much for your reply!
Recordings
Hi,
I'm looking for a GUI like ARI by LittleJohn Consulting (which is not being
maintained actively anymore, but FreePBX seems to include it) so users can
login, check cdrs, recordings, call forward, etc.
Does anyone know of any such working app that can be integrated into vanilla
Because that would mean changing the entire vanilla framework with over 200 users on it.
Original Message
Subject: Re: [asterisk-users] OT: ARI
From: "David Backeberg" [EMAIL PROTECTED]
Date: Mon, September 08, 2008 8:00 pm
To: "Asterisk Users Mailing List - Non-Commercial
David,Please don't mind me needing a GUI just for one purpose of vanilla Asterisk. I do have a separate box, which actually runs PBXiaF and I like it a lot.However, when a project that big in vanilla has been going on, moving on to something else makes no sense, atleast for now.So, either you
in your sip.conf.
One, set "canreinvite" to "no" to keep Asterisk in the call path, that
way it can intercept the DTMF tones. Or, two, set "dtmfmode" to
"info", so that DTMF tones are converted to SIP INFO messages, which
Asterisk will see.
At least, that's h
, 2008 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What do you get when you type show features?
On 9/6/08, Mark Hamilton [EMAIL PROTECTED] wrote:
Hi James,
Thank you very much for a detailed reply. (Matt, sorry about
] On Behalf Of Mark Hamilton
Sent: August 31, 2008 4:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin()
I've tried the regular, xfer button on xlite, dial 100 (to transfer to the
queue), and hit go back to line 1 and hit xfer again
-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What did you try and how did it fail? Are you using the t option in queue?
On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote:
So, no answers or is this thread going to remain unanswered too?
From: [EMAIL PROTECTED
So, no answers or is this thread going to remain unanswered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008 6:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin
Hi,
I'm planning on migrating someone who uses a very mature system. They would
be logging in either as AgentLogin() or AQM. The main requirement however,
is:
The supervisor will have a control panel, where he will see how many of his
agents are on call. If they are, he can right-click on the
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call monitor/barge/train
Mark Hamilton wrote:
Hi,
I'm planning on migrating someone who uses a very mature system. They
would be logging in either as AgentLogin() or AQM. The main
requirement
Hi,
I have the same question as:
http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html
..which like all important things was never answered.
How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's
just pure SIP/VoIP.
Help please.
Thanks,
Oh, by the way, the agent who will be doing the assisted transfer will be
using eyebeam.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008 5:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Transfers
I've asked this question myself, numerous times. But just like a few things
that no one likes to answer to, but are aware of - this gets lost in the
mails without any answers.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: August 20,
I love the thin client stuff. It probably looks as big as the Samsung
SWA-4000. But in terms of hardware, don't I need a PCI card to get it
working? How would that work?
Sorry, I have no idea about Asterisk working for home, but just SIP related
stuff. :(
-Original Message-
From: [EMAIL
Doesn't Queuemetrics run on a license basis?
Anything else that's probably open source and free?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan
Sent: August 13, 2008 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
used them.
regards,
Drew
Mark Hamilton wrote:
I love the thin client stuff. It probably looks as big as the Samsung
SWA-4000. But in terms of hardware, don't I need a PCI card to get it
working? How would that work?
Sorry, I have no idea about Asterisk working for home, but just SIP
related
Hi,
I'd like to install Asterisk at home. But don't want to use a full blown PC
to host it. I was thinking of using fitPC www.fit-pc.com to do all the
Asterisk work, interfacing with the local Bell Canada line, and using a SIP
VoIP line as well.
What do you experts think of it?
Thanks,
Hello,
Have any of you had the chance to see what TeleVantage call monitor is?
It basically shows queues, other extensions, calls in waiting, voicemail,
etc.
Does anyone have if there is something out there like it, so each and every
agent can have one of those? And not just a manager?
I agree. In that case people who use includes in their scripts for which they got paid should pay a portion of their pay to the writer of each include they use.
Original Message
Subject: Re: [asterisk-users] (announce) asterisk T.38 gateway
From: Rob Hillis [EMAIL
iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP
Good luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of spectro
Sent: June 30, 2008 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip extension
, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote:
On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote:
LOL, I agree, it _did_ sound a little complicated than to just schedule
a
call in the future. I apologize for not being able to find
] On Behalf Of Gordon
Henderson
Sent: June 19, 2008 4:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Website callback
On Wed, 18 Jun 2008, Mark Hamilton wrote:
Hi,
I have a website where customers enter their phone numbers to be called.
I'd
like them
Hi guys,
So, I was wondering this morning as to who might have the best recording
solution implemented.
When I say best, I mean how they record, convert it to some
low-diskspace-consuming format, and then leave it there, until a web-app
requests it, and then it's changed to wav or mp3 and
build your own pretty easily.
I may someday sit down and actually go back and re-write it to put out
on the net anyone to use...but we shall see.
Kevin
Mark Hamilton wrote:
Hi guys,
So, I was wondering this morning as to who might have the best
recording solution implemented.
When I say
Hi,
I have a website where customers enter their phone numbers to be called. I'd
like them to have to put in information and 'schedule' a call.
1) Call Immediately
2) Call in the next _ minutes
3) Call me tomorrow, same time.
So, Asterisk will pull two variables from
How can they even set such 1234567890 callerIDs anyway?
For example, our inter/intra state calling depends a lot on the callerIDs.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: June 13, 2008 8:20 AM
To: Asterisk Users Mailing List -
a PRI-T1 in the USA, then you can set outgoing CallerID
with just about any carrier.
MATT---
On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote:
How can they even set such 1234567890 callerIDs anyway?
For example, our inter/intra state calling depends a lot on the
callerIDs.
-Original
How come he has it, and he's in Paris! I'm in Toronto, and I don't have it?
:(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: June 16, 2008 4:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
] wrote:
On June 16, 2008 07:22:18 pm Mark Hamilton wrote:
How come he has it, and he's in Paris! I'm in Toronto, and I don't have
it?
Yeah, me too. I even got a mention in the book, but no screwdriver? :-(
-A.
___
-- Bandwidth and Colocation
Ok, now I'm confused.. logger reload or no logger reload? I want the
Master.csv to rotate.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: June 15, 2008 6:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
/log_rot_ast
endscript
}
/usr/local/bin/log_rot_ast contains:
#!/bin/sh
/usr/sbin/asterisk -rx 'logger reload' /dev/null 21
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Hamilton
Sent: Saturday, June 14, 2008 19:05
To: 'Asterisk Users Mailing List
400 class! I'm in! haha
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 15, 2008 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] World Cheapest Predictive Dialer!
Your original
This sounds good. Except I'm a little confused. Is this a reboot bar which
uses Ethernet to do the reboots? Like a reboot bar, except in a PoE
lifestyle?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 15, 2008 11:36 AM
To:
rotation
While maybe a little too non-cli for some folks, I like to use
Webmin's GUI for setting up log rotation. Nice, quick, easy, hard to
get it wrong
Thanks,
Steve Totaro
On Sun, Jun 15, 2008 at 11:48 AM, Mark Hamilton [EMAIL PROTECTED] wrote:
Yup, drive. Or in Gavin's case Fly.
Really
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] *OT* DLI Ethernet Power Controller $289 (I
paid $200 for a two port webswitch)
On June 15, 2008 12:11:13 pm Mark Hamilton wrote:
This sounds good. Except I'm a little confused. Is this a reboot bar which
uses
as possible.
Thanks,
Steve T
On Sun, Jun 15, 2008 at 12:11 PM, Mark Hamilton [EMAIL PROTECTED] wrote:
Would that help the rotation of Master.csv too?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 15, 2008 11:51 AM
To: Asterisk Users
That works, thanks a lot!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: June 15, 2008 12:35 PM
To: Asterisk Users
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
Mark Hamilton schrieb:
I'm attempting to use this now
Of Gavin Henry
Sent: June 13, 2008 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cdr-custom/Master.csv rotation
2008/6/13 Mark Hamilton [EMAIL PROTECTED]:
Hi,
How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date?
Logrotate
Hi Dean,
Could you please tell me the source of information for your 2nd paragraph?
I'd like to read up more.
Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: June 13, 2008 7:55 AM
To: Asterisk Users Mailing List -
I'm certainly learning a lot from this thread, especially from Steve Totaro.
If only this was OT, I'd love to see a big fat discussion go on regarding
this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: June 13, 2008 9:42 AM
To:
Hi,
How can I rotate /var/log/asterisk/cdr-custom/Master.csv nightly by date?
Thanks,
Mark
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Isn't this exactly why lists such as asterisk-users evolve? I don't see
anything wrong with his question, and I see full connection to Asterisk to
boot! Nothing OT about it either.
So, why, if I may, does he have to ask a smart question, and/or advertise on
asterisk-biz? Did the 'step by step'
in the next is * Really Good Thread
?
Mark Hamilton wrote:
Same here.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: May 22, 2008 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Welcome to the club.
http://lists.digium.com/pipermail/asterisk-users/2008-May/212281.html -
Another similar issue.
http://bugs.digium.com/view.php?id=12709 - Bug report for it.
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carles Pina i
Estany
?
Mark Hamilton wrote:
Same here.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: May 22, 2008 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI
I get this once a week myself.
Original Message
Subject: Re: [asterisk-users] excessive bounces???
From: "Michael Graves" [EMAIL PROTECTED]
Date: Sat, May 24, 2008 11:32 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
[EMAIL PROTECTED].com
I got this also.
In
Also what do I do if I see deadlocks all over the CLI?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: May 23, 2008 12:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping
:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: May 23, 2008 11:17 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
havoc.
Also what do I do if I see deadlocks all over the CLI?
-Original Message
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do anything,
infact all action on CLI stopped.
Then, I did a reload. Same thing.
After that there was no other way.. because even stop now wouldn't work, so
PROTECTED] On Behalf Of Steve Totaro
Sent: May 22, 2008 3:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and causing
havoc.
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote
] reload stopping EVERYTHING on CLI and causing
havoc.
You can create queues via AMI although it won't survive a restart.
Thanks,
Steve Totaro
On Thu, May 22, 2008 at 3:25 PM, Mark Hamilton [EMAIL PROTECTED] wrote:
Thanks Steve.
Using 1.4.18, and I don't think it's fixed.
The unresponsiveness
at 3:38 PM, Mark Hamilton [EMAIL PROTECTED] wrote:
That's true. But that's exactly why it's not used and conf is much easier.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: May 22, 2008 3:34 PM
To: Asterisk Users Mailing List - Non
, 2008 at 3:58 PM, Mark Hamilton [EMAIL PROTECTED] wrote:
1) rains down bullets
2) more work, but surely doable
3) yes, but that's a work around to the problem and inconvenient for 24/7
production box.
Either way, the reload worked perfectly fine before - even with callers
logged in and in full
Totaro wrote:
On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Steve Totaro wrote:
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried
havoc.
BJ Weschke wrote:
Sherwood McGowan wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
anything, infact all action on CLI stopped.
Then, I did a reload. Same thing
Hi Sherwood,
I've done the backtrace. Maybe you can submit yours too.
http://bugs.digium.com/view.php?id=12709
Thanks,
Mark.
Original Message
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and
causing havoc.
From: Sherwood McGowan [EMAIL PROTECTED]
Date: Thu,
Hi Nicolas,
Thank you so very much for this!
(Also on behalf of a large group of Asterisk queue users, I'm sure!)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño
Sent: May 16, 2008 7:50 PM
To: asterisk-users@lists.digium.com
Subject:
Martin,
That's a wicked visualization tool!
Thanks for this contribution!
Thanks,
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin B.
Smith
Sent: May 17, 2008 2:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Lee,
You should probably clean it up and put it up on the wiki. I don't think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it will
be available to everybody.
Thanks for this, btw.
Mark.
-Original Message-
Doesn't he mean something like when the recording happens, he'd like to go
http://192.168.1.1/recordings and then when he sees the list of *.gsm
recordings, he clicks on it, and the serverside starts playing it?
I think you'll need a Quicktime client (as as plugin to your brownser) on
your PC
Since, we're on the the topic of phones, and TFTPing.. if someone on this
thread has some knowledge of putting configs on Cisco IP Phone 7960, can
they please contact me off list?
I've done the configs via tftp, etc but anything into the speaker/handset
relating to voice doesn't work.
Hello,Colors in the CLI have helped me ignore notices/warnings, etc and concentrate better on stuff that I want to look out for during testing.I've noticed that a simple restart would not bring back the colors in the CLI. Asterisk needs to either start on boot, or start by way of safe_asterisk to
Hello,
Colors in the CLI have helped me ignore notices/warnings, etc and
concentrate better on stuff that I want to look out for during testing.
I've noticed that a simple restart would not bring back the colors in
the CLI. Asterisk needs to either start on boot, or start by way of
In article
[EMAIL PROTECTED],
Mark Hamilton [EMAIL PROTECTED] wrote:
Colors in the CLI have helped me ignore notices/warnings, etc and
concentrate better on stuff that I want to look out for during testing.
I've noticed that a simple restart would not bring back the colors in
the CLI. Asterisk
+1 please Thanks!
Original Message
Subject: Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet
From: Darren Wiebe [EMAIL PROTECTED]
Date: Mon, May 05, 2008 6:51 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED].com
If you're willing to cc
I could definitely see a use for such a tool.
Original Message
Subject: Re: [asterisk-users] Manual Wardialer
From: "Steve Totaro" [EMAIL PROTECTED]
Date: Sun, April 27, 2008 11:37 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
[EMAIL PROTECTED].com
Legalities
Steve,
Is this 'shell script' on the public domain? As it sounds really useful. :)
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: April 13, 2008 7:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
/2008, Mark Hamilton [EMAIL PROTECTED] wrote:
I find it hard to believe no one knows, so is it just plain no helping? J
If someone would like to atleast point me in the right direction that will
deal specifically with what I'm asking, that would be appreciated too.
Much thanks.
From: Mark
I find it hard to believe no one knows, so is it just plain no helping? J
If someone would like to atleast point me in the right direction that will
deal specifically with what I'm asking, that would be appreciated too.
Much thanks.
From: Mark Hamilton [mailto:[EMAIL PROTECTED]
Sent
Hello,
I'm a little confused on DTMF.
A sip peer is registered on two Asterisk servers. No dtmfmode is set for
them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
register on each other.
A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call
is
Does this announce only on the user's side or the other side as well?
I mean if a user is on call with a customer, and you do the system announce,
will the announcement be heard by the customer as well? As that would be
bad, heh.
However, I'm really interested in checking it out. So, please
Hello,
I have two sites. Both the sites will require queues for their own reason,
own campaigns, etc. Like site1 would handle product1, product2, product3,
while site2 can do product4, but can also do customer support for product1
and if anything, can transfer to site1's product1 queue
I'm confused.
Is this spam, or what, really?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz,
Steven
Sent: March 24, 2008 2:56 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FYI about my Mona Vie business venture
Asterisk work
I would think you'll need to do a Playback() of this message before the
caller enters the queue, as I'm not aware of such an option provided by
app_queue.
Exten=100,1,Answer()
Exten=100,n,Playback(greetings-earthling)
Exten=100,n,Queue(xyzqueue)
Exten=100,n,Hangup
-Original Message-
I've been using Druid since their V since their V.3.2.5, and I have to say I
love it. I haven't done a lot of advanced things with it though, but I know
that their new version has almost all of it. Upgrade is not an option, a new
license is, which is why I'm still with V325.
Either way, Druid is
Lee,
I'm pretty sure you can using macros and what not. Unfortunately, I'm not
that experienced to comment on it, but I'm positive that can be done with
one of those if, else things.
Mark.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee, John
Hello,
Asterisk Server A makes an outbound call, and upon connect:
exten
=1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT
)
(${connectto} most of the time happens to be [EMAIL PROTECTED] or 54321 {IP
masqueraded ofcourse})
..transfers it to * Server B (i.e
Hello,
Just got a core dump, and thanks to somebody named Matt, I ran a gdb
command, and this is the cusp of it:
Reading symbols from /lib/libgcc_s.so.1...done.
Loaded symbols for /lib/libgcc_s.so.1
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with
Hello,
Didn't get much help on asterisk-dev, so here it is.
Please help.
Thanks.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: March 13, 2008 2:10 PM
To: [EMAIL PROTECTED]
Subject: [asterisk-dev] Hardware and CentOS tweaks.
Hello,
We're
Help needed, please.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: March 13, 2008 2:16 PM
To: [EMAIL PROTECTED]
Subject: [asterisk-dev] Call failed, reason 0 explanation.
Hello,
I understand that Asterisk interprets SIP error codes as 'reason
I don't think the link that Lee gave works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: March 14, 2008 5:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Logs for Call generated by
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