If you have automated the configuration process, all you have to do is:1) Set option P75 (Daylight savings time) to 0 or 1 accordingly in the configuration file.2) Regenerate the compiled configuration file(s).3) Make sure they are in the TFTP or Web server.4) Reboot the phones so they read the
You can get wget for OSX from DarwinPorts (http://wget.darwinports.com/)
On Oct 17, 2006, at 4:13 PM, Martin Joseph wrote:
On 2006-10-17 01:06:25 -0700, Tzafrir Cohen
[EMAIL PROTECTED] said:
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote:
SVN Trunk doesn't currently build
We can do attended transfers on the GXP-2000 just fine with a single
account.
When you have a call on Line 1, simply press Line 2 (Line 1 will be
put on hold automatically) and press SEND. Once the other party picks
up, you announce the call and then press TRNSFR and then press Line 1.
-
I have a couple of clients with a bunch of GXP-2000. They can do
attended transfers with no problems. However, there are times that
the party to transfer to is simply not at their desk and the party
wanting to transfer the call knows that. In these cases, they'd like
to blind transfer the
You need to enable call waiting on the phone's config.- DanielOn Sep 8, 2006, at 9:35 AM, Zeeshan Zakaria wrote:First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, "all lines are busy, please
Then back to default.
Ring_WaveformSinusoid/Ring_Waveform
PAP2-NA shouldn't be any different.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: Friday, August 25, 2006 6:27 PM
To: Non-Commercial Discussion Asterisk
Subject
I have a few PAP2-NA that are being mass configured using the
instructions on the wiki for the Sipura mass configuration.
However, I need to make sure the following settings are in place as
follow:
Under the Regional Tab, I need the Ring Waveform to be Trapezoid
instead of Sinuzoid and
I have a client with about 24 GXP-2000. Everything seems to be
working fine except one particular behavior of the blind transfer.
Whenever anyone makes an outbound call, they can transfer the call
between extensions either blind or attended with no problems.
However, whenever an incoming
Is there anyway to reach ANYONE on the phone (in the US) at 4PSA?
Every time you call their office, you just need to leave a message.
It prompts you to enter an extension (if you know any), but whenever
you press ANY digit, it simply goes straight into voicemail. Is this
company for real?
I have an Perl AGI script which accepts inbound calls and offers an
IVR service. Depending on certain options that are selected on the
IVR, the script is supposed to dial-out an external number, and
therefore, basically, conference the original caller with an external
number. That part is
That is exactly what I'm looking for. Thanks,
Daniel
On Jul 31, 2006, at 9:23 PM, Moises Silva wrote:
may be you are looking for asterisk application ForkCDR(), more info
in voip-info.org
Regards
On 7/31/06, Daniel Salama [EMAIL PROTECTED] wrote:
I have an Perl AGI script which accepts
, Daniel Salama wrote:
That is exactly what I'm looking for. Thanks,
Daniel
On Jul 31, 2006, at 9:23 PM, Moises Silva wrote:
may be you are looking for asterisk application ForkCDR(), more info
in voip-info.org
Regards
On 7/31/06, Daniel Salama [EMAIL PROTECTED] wrote:
I have an Perl AGI script
Looking for a SIP or IAX softphone for a call center application that
can do G729 codec. Any recommendations? Ideally it would do screen
pops, meaning that it will understand the URL option of the Dial
command.
Thanks,
Daniel
___
--Bandwidth and
I have the eyeBeam softphone but I don't see G729 in the list of
available codecs (BTW, this is the paid version not X-Lite). Any clues?
Thanks,
Daniel
On Jul 24, 2006, at 12:00 PM, Guillermo Salas M. wrote:
On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote:
Looking for a SIP or IAX
Check the default context defined in zapata.conf which is where
incoming calls will go to. It may be going to a context that you are
not aware of.
- Daniel
On Jul 4, 2006, at 3:46 AM, Pierre du Plessis wrote:
Greetings,
I have installed a new FXO card but even though there's no incoming
I'm trying to setup an Asterisk box as an H323 to SIP gateway.
Basically, I'd like to receive traffic in H323 and forward to another
Asterisk box (on the same network) using either IAX2 or SIP so that
the second Asterisk box communicates with other gateways using SIP.
Therefore, if I
If pricing is an issue, I've had very good experience with GXP-2000. Otherwise, I really like the SPA-941/2.- DanielOn Jun 28, 2006, at 3:05 PM, Forrest Beck wrote: We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to
I have an Perl AGI script that acts as an IVR for my Asterisk box.
Basically, it simply plays audio files to the caller, collecting DTMF
input and logging every DTMF input into a database table, simply to
document every step or option selected by the caller.
One thing is that in addition
Beautiful. Will test and give you comments.
Nice work.
- Daniel
On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote:
Daniel Salama wrote:
Dustin,
any updates on this?
Thanks,
Daniel
Hey Daniel!
Yes - just posted the link.
I appologize for the delay.
Here's the link to the forum as well
Dustin,
any updates on this?
Thanks,
Daniel
On Jun 23, 2006, at 1:07 PM, Dustin Wildes wrote:
shadowym wrote:
That feature is called Bridged (or Shared) line appearance. That
is one of
the things Asterisk cannot do and nobody seems very interested in
making it
do that because it is
I had the same problem some time ago. Make sure call waiting is NOT
disabled. This will make the phone receive more calls on the other
lines.
- Daniel
On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne
wrote:
I have a network of GXP 2000 phones and would like to know if
I have a client with 20 GXP-2000s. Everything seems to be working
fine. However, after a couple of weeks of use, the client is having a
hard time adjusting to the new IP based phone systems and only misses
one feature from their old Lucent system.
That is, they had 8 analog lines before
I've read in different places that if I want to do trunking and
meetme on Asterisk I need to have a reliable timer. People have
recommended that I install a Digium board, even if I don't have any
circuits connected to it, just to get a reliable timer. However, I've
also read that if I'm
Given that the NSLU2 can't do trunking, do you think that a PIII
733Mhz, 128MB RAM will do?
Thanks,
Daniel
On Jun 15, 2006, at 4:15 AM, Tim Panton wrote:
On 15 Jun 2006, at 02:59, Daniel Salama wrote:
Does anyone know how many simultaneous calls can a WRTG54GS
handle? Assuming SIP
Is there anyone that could explain to me the phenomenon of Echo or at
least point me where I can learn more? Why is this affecting the VoIP
world so much and not the regular PSTN analog world? What does the
PSTN industry have that they can handle such high volume of calls and
there is no
I have been reading about integrating Asterisk with SER to help
Asterisk deal with large volume of registrations (mainly). I was
planning on fronting Asterisk with SER for that purpose. Not that I
have the traffic at this moment, but because I wanted to get the
infrastructure in place.
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random calls
so as to keep an updated database on a per provider, per destination,
per time-of-day score? Hopefully, with that information we can
at 01:26 -0400, Daniel Salama wrote:
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random calls
so as to keep an updated database on a per provider, per destination,
per time-of-day score? Hopefully
It sounds nice, but, how many calls can you get on the NSLU2? Say the
SIP phones are talking either G711.u or GSM only and the IAX trunk is
GSM only.
Thanks,
Daniel
On Jun 15, 2006, at 4:15 AM, Tim Panton wrote:
On 15 Jun 2006, at 02:59, Daniel Salama wrote:
Does anyone know how many
Wow! Can anyone comment on this? If this was the original suggestion,
can anyone confirm that trunking DOES work on the NSLU2?
Thanks,
Daniel
On Jun 15, 2006, at 10:47 AM, Kristian Kielhofner wrote:
Daniel Salama wrote:
It sounds nice, but, how many calls can you get on the NSLU2? Say
On Wed, 2006-06-14 at 04:17, Daniel Salama wrote:
I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u
codec, I know they upstream bandwidth
.
A better alternative is to get them to upgrade the DSL to 512 uplink.
Tim.
On 14 Jun 2006, at 17:11, Daniel Salama wrote:
Wow! 22Kbps of overhead? Are you sure? That sounds like way too
much overhead. I can't use IAX2 because the GXP-2000 are SIP
phones :( Any other suggestion?
Thanks,
Daniel
Can anyone explain to me what this means:
Jun 14 19:46:10 NOTICE[7391]: rtp.c:331 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: 66.175.1.1
When I try to make a call from certain IP phones, I see that message
on
. It is
there to keep you from hearing no sound through the speaker and
thinking you have been disconnected.
Check your phone's config for comfort noise or silence suppression
and turn it on or off respectively.
What phone model(s) do you see this with?
Daniel Salama wrote:
Can anyone
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and the
WRTG54GS connects to a remote Asterisk server using IAX2 trunking
using GSM codec.
Thanks,
Daniel
___
Daniel Salama wrote:
They are extremely casual web surfers. Just have their Outlook
client opened checking email every minute. Email traffic is very low.
They are all connected to the same switch. It's a Netopia DSL
router/modem/switch for the BellSouth DSL service. The computers
I have a client with about 16 GXP-2000. They complain that the audio
quality is terrible after 2 or 3 simultaneous conversations. They are
behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u
codec, I know they upstream bandwidth is the limiting factor and they
most likely
This seems to be only for prepaid calling cards. Is there something that can also handle prepaid and postpaid multi-tenant SIP phones?- DanielOn Jun 12, 2006, at 5:00 PM, William Piper wrote:www.asterisk2billing.org On 6/12/06, Wasif [EMAIL PROTECTED] wrote: Hi,I need to use Asterisk as a switch
.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?
Thanks,
Daniel
___
--Bandwidth
in the
HTML are the same as the ones that go in the config file.
p.
On Sat, 2006-06-10 at 02:06 -0400, Daniel Salama wrote:
Wow! Awesome. This template is much more complete than the one on
GS's download page.
Thanks,
Daniel
On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote:
Yes you can as long as you
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file? If so, what are the
parameters to put in the configuration file?
Thanks,
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --
Is there any open source software capable of managing Asterisk to
offer Virtual PBX services to multiple clients, including billing? Or
is there a combination of open source initiatives that offer this?
Thanks,
Daniel
___
--Bandwidth and Colocation
While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000.BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice.Thanks,DanielOn Jun 7, 2006, at 8:09 AM,
They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another
Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648]
The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically
Latest firmware installed and problem with handset. They don't use
headset nor speakerphone.
Thanks,
Daniel
On Jun 7, 2006, at 3:14 PM, John Novack wrote:
Daniel Salama wrote:
snip
As for the SPA-841, I have a client with a few of them and he
cannot stop complaining about the bad
. That is not the
fault of the phones. Are you sure you didn't change anything else
when you switched from the spa-841 phones?
Daniel Salama wrote:
The complete opposite. The user complaints that either they cannot
hear the remote party well or the remote party cannot hear them
well. Sometimes
:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7
I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems
to be working fine. However, there are a couple of issues I'd like to
know if are possible:
1) Even though the phone has 4 line appearances, if I am speaking on
a line, the phone can no longer receive phone calls. I can
release firmware, not a beta.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: June 6, 2006 4:12 PM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] GXP-2000
I'm using a few GXP-2000 with firmware 1.0.2.13
with Polycom's as new people come on and as others just get fed up. Unfortunately one of the phones met it's doom by way of a hammer. But I guess, what do you expect for under a hundred bucks. Erick On 6/6/06, Daniel Salama [EMAIL PROTECTED] wrote: I enabled call-waiting from the tftp
Lacy,I am in a similar situation, except that my users are extensions aware. However, I'd love to know how you solved your problem since call transfer seems a bit complicated at the moment.Thanks,DanielOn Jun 2, 2006, at 6:51 AM, Lacy Moore - Aspendora wrote:Kerry, so to park a call, you would put
Does FreePBX support virtualization of its services? For example, can
I use it to provide virtual PBX to different clients under the same
instance of FreePBX? Or is it more geared to single office-type
installation?
Thanks,
Daniel
___
--Bandwidth
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: Thursday, May 25, 2006 10:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] FreePBX virtualization
Does FreePBX support virtualization of its services? For
example, can
PROTECTED] On Behalf Of
Daniel Salama
Sent: Thursday, May 25, 2006 10:42 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] FreePBX virtualization
Does FreePBX support virtualization of its services? For
example, can I use it to provide virtual PBX to different
clients under the same
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Daniel Salama
Sent: Thursday, May 25, 2006 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FreePBX virtualization
Any alternate open-source solutions?
On May 25, 2006, at 2:17 PM, Douglas
like you don't have USB support compiled in the kernel.
Chris
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 11:55 PM
Subject: Re: [Asterisk-Users] Problem
Can anyone provide more information on switch or point me to where I
can find more about it?
The only I've been able to find on the wiki is:
http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers
and towards the bottom of (section Forwarding to another Asterisk):
Adam,
See my comments below:
On May 12, 2005, at 10:12 PM, Adam Goryachev wrote:
On Thu, 2005-05-12 at 10:35 -0400, Daniel Salama wrote:
What I have discovered is that my motherboard only supports usb-ohci
and not usb-uhci. Reading on the wiki, it says that ztdummy requires
usb-uhci.
To make
I'm noticing by watching the CLI that my inbound calls coming via T1s
on a TE410P are using GSM codec. Why wouldn't it use ULAW as default?
How can I make it use ULAW as default?
Thanks,
Daniel
___
Asterisk-Users mailing list
I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm
getting the following problem:
-- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to
open
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel
Salama
Sent: Wednesday, May 11, 2005 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with MeetMe
I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm
getting
the wcfxo, wcfxs, or ztdummy loaded?
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 1:09 PM
Subject: [Asterisk-Users] Problem with MeetMe
I'm trying
I just came across http://www.voip-info.org/tiki-index.php?
page=Asterisk%20TDMoE and seemed very interesting. It prompted me to
question whether it would be more efficient to do TDMoE or IAX2.
The application is very simple. I have two asterisk boxes. One is
strictly a gateway to the PSTN
USB for
ztdummy to load.
Chris
- Original Message -
From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 11, 2005 3:28 PM
Subject: Re: [Asterisk-Users] Problem with MeetMe
I don't have any
I just installed a TE410P on a Debian Sarge system running kernel
2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine.
However, I have a couple of problems with the TE410P and Zaptel.
First, the TE410P is showing me red alarms on 2 of the 4 T1s. This
board (the TE410P) was just
Is it possible to set a variable for an IAX device in iax.conf that
can be read from the dial plan (extensions.conf)? If so, can you
explain?
Thanks,
Daniel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
you can create globals like:
JOHN=IAX2/1234
And then use ${JOHN} in your dial plan to use that device.
Alfredo.
On 5/10/05, Daniel Salama [EMAIL PROTECTED] wrote:
Is it possible to set a variable for an IAX device in iax.conf that
can be read from the dial plan (extensions.conf)? If so, can you
This is a good solution. However, I have a regarding this approach:
Does this mean that ANY incoming directed to that agent will fall
into that context? I have calls coming into the system and being put
in a Queue. When the agent is available, the call will be de-queued
to the agent. When is
Sorry for the confusion.
On May 10, 2005, at 10:45 PM, Tzafrir Cohen wrote:
On Tue, May 10, 2005 at 02:52:39PM -0400, Daniel Salama wrote:
I just installed a TE410P on a Debian Sarge system running kernel
2.6.11-1-686-smp.
Which is a kernel from sid, actually. What version is it exactly?
dpkg -l
I'm setting up a demo for two asterisk machines. One will be a
central Asterisk server which will handle everything already in VoIP
(office-like functions plus agents functionality). The second
Asterisk box will be used strictly as a VoIP gateway to the first
server.
The gateway server
And how/where can you download the latest firmware onto /var/lib/
asterisk/firmware/iax?
Thanks,
daniel
On May 7, 2005, at 9:13 AM, Time Bandit wrote:
I'd like to known what I have to do to upgrade
the firmware into a IAXy device.
It does it automagically when it connect to Asterisk if a newer
Not entirely sure, but, I wonder what would happen if you define RING
in the [globals] section first, and the use SetVar or SetGlobalVar in
the other contexts to override its value.
- Daniel
On May 8, 2005, at 2:48 AM, Mark Wormgoor wrote:
Hi,
Is it possible to set a variable for a context
Question: can you override the caller id at all for outbound calls
(whether statically or on-the-fly)? Remember that you need to have the
carrier give you the ability to override the caller id. This is
normally called digit manipulation and is normally regulated by the
carriers so as to avoid
I've read on the wiki that the Asterisk Database is mainly (or only)
used for manipulating the dial plan (e.g. extensions.conf). I know that
RealTime can be used for much more than that (i.e. sip.conf, iax.conf,
etc).
My question is: if I had two management applications, one that uses
.
regards
- Original Message - From: Daniel Salama [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 04, 2005 4:21 AM
Subject: [Asterisk-Users] Hardware Capacity/Configuration
I know this is a frequent topic
I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and
I'm just noticing that the TE410P does not fit in the PCI slot. It
seems as if the little opening in the PCI is on the wrong side. Has
anyone else seen this or is it just me and I'm too stupid to do
something as basic as
are *definitely* 5v
slots.
--Rob
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Daniel Salama
Sent: Thursday, May 05, 2005 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] TE410P does not fit in motherboard
I'm trying to setup a multi-tenant configuration of * and have the
following question:
In extensions.conf, there is a [global] section that I would normally
use to define global variables for my single tenant setups. Now, is
there a way to have something like global variables on a per tenant
[incoming]
Exten = 1234etc...
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005 6:50 PM
To: Tim Connolly
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP over IAX2
I understand and I guess I know
Sorry if this is posted twice. I sent this about an hour ago and haven't seen it in the list yet.
Thanks,
Daniel
Begin forwarded message:
From: Daniel Salama [EMAIL PROTECTED]>
Date: May 3, 2005 1:12:51 PM EDT
To: Tim Connolly [EMAIL PROTECTED]>
Cc: 'Asterisk Users Mailing List - Non-Comm
I'm trying to configure 4 T1s into this board. The T1s work just fine.
However, I have a question about setting up the clock source properly.
3 T1s are from the same carrier and the remaining T1 is from another. I
have a configuration similar to:
/etc/zaptel.conf
span=1,1,0,esf,b8zs
em=1,24
I know this is a frequent topic on the list. Sorry if this creates more
bandwidth but I couldn't get my specific answer from neither the wiki
nor searching the list.
I've read that a P4 3GHz+ should be sufficient to handle a 4 T1s on a
single CPU machine. I am setting up a proof of concept
I've read on the wiki how you can SNMP monitor an Asterisk machine and
from what I read, you're pretty much monitoring the availability of
Asterisk.
I'm looking for a way to be able to monitor the availability of
individual T1 circuits of my TE410P card. During the storm season, some
of our
Hi, I recompiled Asterisk 1.0.3 on a machine which I upgraded the
kernel. I also recompiled zaptel and libpri.
After doing this, I am realizing that I'm having some problems playing
mp3 files. However, and very strangely, music on hold is working
playing mp3 files.
I have an AGI script that
Is there a way to configure asterisk to execute an AGI script upon the
transferring of a call to an extension from the Queue? For example,
once the call is put in the queue and the extension becomes available,
the Queue app will send the call to that extension. Is there a way for
me to
Anyone know if Digium cards, especifically TE410P, are compatible with
BSD (FreeBSD or NetBSD)? How does * run on BSD?
Thanks,
Daniel
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
the sirections, and the card i
believe does work
On Tue, 2005-05-03 at 00:01 -0400, Daniel Salama wrote:
Anyone know if Digium cards, especifically TE410P, are compatible with
BSD (FreeBSD or NetBSD)? How does * run on BSD?
Thanks,
Daniel
___
Asterisk-Users mailing
Along the same lines, is there some sort of capacity chart that maps
capacity based on translations/transcoding?
- Daniel
On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote:
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw?
I have two asterisk boxes. I'm running an IVR script in one of them and
I have agents registered on the second box.
I wish to create an extension on the * box where the agents are
registered, so that when dialed, it will connect the agent to the IVR
script on the other * box. However, I'd like
I was reading on the wiki about the supported kernels and I __THINK__
the main issues with the kernel versions have more to do with Zaptel
driver and not necessarily Asterisk itself. Is this correct?
Thanks,
Daniel
___
Asterisk-Users mailing list
to dial the extension on the IVR, you should be okay.
Just
make sure the default SIP context on the IVR has that extension
defined, or
define the IVR box as a SIP peer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Salama
Sent: Saturday, April
canreinvite=yes
qualify=yes
extension.conf
[incoming]
Exten = 1234etc...
-Original Message-
From: Daniel Salama [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 30, 2005 6:50 PM
To: Tim Connolly
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP over
recording calls?
Thanks,
Matthew Roth
http://voip-info.org/tiki-index.php?
page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
Thank you again. I will definitely do that. By cheaper asterisk
servers, do you mean single-CPU machines that can handle Quad T1s and
still do the call monitoring
Well, I don't think I'm ready to spend that much money :)
I understand your point regarding that load depends on usage.
SIP_Agents are simply agents answering calls. Average call length would
be about 8 minutes. During some of these calls (maybe 25%), agents will
conference the call (PSTN
I think that would be a great idea. The only problem I see is that
Asterisk is growing its feature set and maturing at such a dynamic
rate, that I don't know in many cases, where to point the finger at.
Sometimes it's stability of the CVS version, sometimes it's stability
of Digium or
This is an interesting question. I haven't tested it but would love to
know if it works or not. Anyone?
- Daniel
On Apr 29, 2005, at 3:38 AM, Michael Welter wrote:
I haven't seen this before--can an agent log into a queue on a remote
(i.e. over IAX) Asterisk server?
Does anyone have any experience with servers from siliconmechanics.com?
Are they reliable? How does * run on them?
Thanks
- Daniel
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com. At
least companies like that really know their
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and
then we hang up. We have about 100 DIDs routed to different contexts
and I wouldn't want to have to
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