trying to stay on top of a complex and diverse environment like a
PBX.
--
Zach Segal
Network Operations
Gleim Internet
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
resolves things however this is
not an option in the middle of the day. Thanks in advance for all your
help.
--
Zach Segal
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
For what it's worth, Apple's Mail application automatically embeds a tiny QuickTime Player interface in mail messages that contain audio attachments. The result looks like this:http://zachfine.com/blog/images/voicemail_sample_in_apple_mail.gifI'm sure it would not be too difficult to embed a small
camp in
order to pass the certification test? The boot camp course outline seems
very basic and I know much more than that. Please advice me what should
I do for the preparation for this certification's test.
Zach
___
--Bandwidth and Colocation provided
How different is FreePBX from Asterisk @ Home?
Zach
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
your prompts, what will you do with the voicemail, directory and
some other system prompts? Or youll need to
change all the required sound files too to make all the voices consistent.
Zach A
___
--Bandwidth and Colocation provided
Kernel is 2.6. zaptel and ztdummy load with Linux, so I can check in
lsmod that they are loaded. They load without any problem, I've loaded
them manually too.
Zach A
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Friday, March 10, 2006 6:13 AM
To: Asterisk Users
Kernel is 2.6. zaptel and ztdummy load with Linux, so I can check in
lsmod that they are loaded. They load without any problem, I've loaded
them manually too.
Zach A
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Friday, March 10, 2006 6:13 AM
To: Asterisk Users
,
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
What port does mpg123 uses to play music on when it starts MoH after
asterisk has put called on hold?
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
Hi,
Is it possible to do this in extensions.conf to put a caller
in queue and dial an agents extension so that he knows that somebody is
in queue waiting to be answered. This agent will be a remote agent and
extension will dial his cell phone.
Thanks
Zach
I found this link useful for SPA3000
http://www.darksource.org/index.php?option=com_contenttask=viewid=43I
temid=2
Zach A
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Monday, March 06, 2006 6:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
? Or is there any other way of receiving faxes. Can SPA3000
send fax calls directly to the fax machine when detecting the short
rings? I need some solution to receive faxes.
Thanks,
Zach A.
___
--Bandwidth and Colocation provided by Easynews.com
SoX needs that libid3tag, libmad and madplay are installed before it can
read mp3 files and convert them into some other format.
Zach A
-Original Message-
From: Chris Stenton [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 02, 2006 3:30 AM
To: Asterisk Users Mailing List - Non
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
Hi everybody,
It should be a simple thing to do but I
dont know how to do it. Now I have 2 DIDs and
I want one of them go to [context1] and other one to go to [context2]. How can
I achieve this.
Thanks,
Zach A
___
--Bandwidth
=xxx.xxx.xxx.xxx
fromuser=DID2
Now both DIDs are sent to context [incoming] which is the default
context for SIP. If I add context=incoming2 under any DID section, it
doesn't go to that context and still go to the default context. How can
I direct DID2 to [incoming2] context?
Zach A
-Original Message
to a
customer? I don't want to call it a PBX.
Thanks
Zach A.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
displays.
Or maybe there is a way for SPA3000 to find the caller ID a littler
faster, as all the other phones do which are directly connected to the
Bell line.
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
If not in spa3k, then how about digium hardware, will that be faster in
picking up caller IDs or is it possible to make it work faster. I need
only one FXS/FXO. Is X101P single FXS/FXO?
Zach A.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 22
Hi,
My telephone extensions on asterisk which itself is connected to the
Bell line using SPA-3000, ring only after third ring from the caller.
Why is this happening and what is the solution?
Thanks
Zach A.
___
--Bandwidth and Colocation provided
,
changing RTP ports but nothing helped. It plays ok only for incoming SIP
calls from the same provider. Any guess why it doesn't work for other
incoming calls and is always choppy?
Thanks,
Zach A.
___
--Bandwidth and Colocation provided by Easynews.com
Hi,
I tried to connect SPA 3000 to bell phone line but it didn't work as it
should. Incoming calls worked but outgoing didn't. I need help on its
configuration. Couldn't find useful information on the Internet so far.
Zach A.
___
--Bandwidth
to use
ARI. What I need to do for that. As I understand I need to install
Apache and MySQL on the same machine. What else I need to do. Is there
any step by step guide about it, or can somebody help me on this?
Thanks,
Zach
___
--Bandwidth and Colocation
Hi all,
I had problem running MySQL on FC3 and what I found from googling was
that SELinux should be disabled to make MySQL work n FC3. Now I am
concerned about Asterisk, is it a good idea to disable SELinux. Or is
there any other way to make MySQL work without disabling SELinux?
Thanks,
Hi everyone,
What is recommended for a production quality system, FC3 or FC4. Once
installed, is it necessary to run yum update, does that make things any
better or just take up more memory?
Zach A.
___
--Bandwidth and Colocation provided
Hi,
Is there any detailed guide/tutorial source online on queues?
Zach
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
There is no good help on wiki and voip-info.org, I've gone through it
already.
Zach
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Monday, February 06, 2006 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help
then?
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Zach A
Skickat: den 6 februari 2006 18:31
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Ämne: RE: [Asterisk-Users] Help on queues
There is no good help on wiki and voip
Thanks johann for your reply and examples. I'll try those and will go
through wiki again.
Zach A
-Original Message-
From: Johann [mailto:[EMAIL PROTECTED]
Sent: Monday, February 06, 2006 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
.
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
.
Zach A
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi everyone,
How can I make a caller go back to the main menu if he gets tired of
waiting in a queue for too long?
Thanks,
Zeeshan A Zakaria
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or
, but output files sound
terrible. And still they played distorted. But other gsm voice prompts
played ok when played as MoH.
What should I do now for MoH. I have everything working perfect except
MoH. Why only music files are distorted and not Allison's voice prompts.
Thanks in advance for help.
Zach
Hi Everybody,
My MoH is distorted when a call comes in from PSTN line or Cell Phone. I
have SIP phone line and SIP to SIP is good. I don't have any digium
hardware installed, instead I am using ztdummy and apparently it is
working without any problem. What could be causing this distorted sound
... hopefully someone else on the list
will have something more.
-- Zach
On Tue, 22 Mar 2005 21:59:54 +1100, Rob Gillan
[EMAIL PROTECTED] wrote:
Hi,
Know there has been numerous posts on the subject of asterisk-addons
and OS X. We have other uses for MySQL on the machine so changing over
Thanks! I took the advice on that website, and by changing the line:
format_mp3.so: $(MP3OBJS)
$(CC) $(CFLAGS) -shared -Xlinker -x -o $@ $(MP3OBJS)
to
format_mp3.so: $(MP3OBJS)
$(CC) $(CFLAGS) -bundle -flat_namespace -undefined suppress
-Xlinker -x -o $@ $(MP3OBJS)
in
... through
several rounds of trouble shooting the only time I could get it to
work was with 0.0.0.0.
If anyone has an idea on how to re-bind successfully, please let me
know! Thanks in advance. If any .conf details would be helpful I'll
send 'em along.
-- Zach
life without those modules (zaptel, libpri, format_mp3, probably
others)?
Thanks,
Zach
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
months ago where some
people had success getting rid of this by using format_mp3. But
unfortunately I can't, since it uses -shared to make. Oh well :-(
On Mon, 21 Mar 2005 16:06:00 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
Zach Scott wrote:
Hey all again,
I have successfully compiled
command. During one freeze I actually got
some kernel oops info but had no way to save it. Working on that now. I
have also tried moving the card to a different PCI slot with no effect.
So, any other ideas are welcome, along with any ideas for obtaining
additional debugging info.
Thanks,
-Zach
these cards
for data-only as opposed to voice?
Thanks,
-Zach.
[EMAIL PROTECTED] wrote:
What motherboard do you have?
http://www.voipinfo.org/wiki-Asterisk+hardware talks about problems with
T100P and some motherboards/chipsets. Mostly ones with shared video memory.
-Bill
-Original Message
in the archives, but no real answers to the problem. Any idea
where to start with this?
Thanks,
-Zach.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
using any of the
fedora kernels which is what drove me to the stock kernel to begin
with. I'll check the archives again on that issue. The chipset is a
VIA chipset. I don't know much about it other than that yet. I'll try
PCI slot move as well.
Thanks,
-Zach
44 matches
Mail list logo