I have had no problems with the Ambient MD3200 I bought
off ebay. It was advertised as an asterisk fxo, i didn't know which chipset
I was getting until it arrived.
Hope this helps,
Jon.
On Tuesday 09 August 2005 01:02 pm, Douglas Logan wrote:
Now that the X100P is no longer being offered by
I assume you mean to save on tolls between the two offices.
If so, the simplest way is to set up asterisk on both ends
and specify in your dialplan which numbers you want to
go out over IP and which you want to go out over landline.
Asterisk makes this easy as it uses the most specific first,
so
the flash operator panel does it great.
You need to use the agent channels instead
of the zap/sip channels.
Try this:
[Agent/101]
Position=3 ; Button number in the console
Label=Steve 101
Extension=101; Extension to reach that channel
Context=localext ; Context where
On Tuesday 12 July 2005 10:36 pm, Allan Regenbaum wrote:
Could someone please help me to understand how to
a) customize voicemail . so that I can say Hi this is Allan, not in
right now .. I have read that one can press 0 to get to the
voicemail menu..but I cant figure at what point
I love the spa-3000 and related devices.
I know they're not IP phones, but they turn any
phone you want into one without sacrificing the
features of either.
Jon.
On Tuesday 12 July 2005 11:16 am, Alexandre Leclerc wrote:
Hi all,
We are in the process of selection IP Phones to work with our
When someone calls for an agent and the agent is
taking an incoming call, the agent is registered as
busy, but if the agent is on an outgoing call while being
logged in with agentcallbacklogin, the agent
is instead assumed to be unavailable. How do I correct
this behavior?
Thanks,
Jon.
Make sure your fxo and fxs are in two different groups.
Otherwise, you won't be able to specify which one to steal.
Also, check out zapbarge, that should work better than meetme
for what you are trying to do.
Hope this helps,
Jon.
On Wednesday 22 June 2005 01:18 pm, Richard Koch wrote:
I'm trying to get canreinvite=yes to work. I would like
asterisk to release the line and let the 2 ports on the sipura
device to talk to each other directly. Is there a setting
I need to activate on the sipura device, or is there something
else I need to do? It's possible that it is a nat
Check out ackcall=yes in agents.conf
It allows them to press # to accept, or press * to not accept.
then you can do something like:
exten = 101,1,Dial(Agent/101,20,A(presspoundtoanswer))
or if you want to get more fancy, check out queues.conf
where you can set ring orders and answer penalties.
Try changing wait(2) to background(silence/9)
See if that helps, as you can't accept digits during a wait.
Also, check out the the waitexten command.
Hope this helps,
Jon.
On Wednesday 25 May 2005 05:14 am, Kamran Ahmad wrote:
hello
like if 6000 is the main exchange number. any one dial
I always use just Dial(Sip/2000,20)
and it has always worked just fine for me.
The below example is not only unneccessary, but is broken.
In the below example, it will dial the sip device until it is answered
or a busy is received and then it will ring for 20 seconds for no
reason before hanging
I'm still looking for a good solution, but
here is a place to start, let me know what you
end up deciding.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20campon
http://www.voip-info.org/wiki-Asterisk+call+parking
use option m in the cmd dial.
Cheers,
Jon.
On Saturday 21 May 2005 03:26 pm, hank smith wrote:
hello how do I set up asterisk to play music on hold to callers while it
rings my phones? I am using the amp portal to configure the asterisk pbx
just to let you all know. thanks
hank
email:
What cvs versions are people using in a production environment?
Is there a good way to choose a cvs version, like maybe one right
before a major change? Are there any plans on having a testing
branch that is not as stable as stable, but not as bleeding edge as
head? I'm thinking of compiling a
The persistentmembers=yes is suppose to keep agents in a queue
over a restart. It might do this, but it doesn't do much good as
even if they all remain in the queue, they are all logged out on a
restart. Is there any way to keep the agents that are logged in, logged
in across a restart?
Thanks, I must have looked at the list of available
commands a dozen times, I knew that there almost
had to be one, but that one kept hiding from me.
Thanks,
Jon.
On Thursday 19 May 2005 02:12 am, Dave Cotton wrote:
On Wed, 2005-05-18 at 16:02 -0500, Jon Gabrielson wrote:
Thanks
Ok, this is probably a stupid question, but I can't seem to find
anywhere where it tells how to put someone on hold on a zap channel.
Flash gives me a dialtone and # tells me to enter a new
extension, how can i just put the caller on hold. Pressing # then
hanging up drops the call. Is there a
function, not
Asterisk. If you go to RadioShack (or anywhere else that sells phones)
you can find analog phones that have a hold button.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jon Gabrielson
Sent: Thursday, May 19, 2005 4:58 PM
I would like the ability for a sipura sip device to
instantly connect to an asterisk server as soon as
the sipura sip device goes offhook and before
any digits are pressed. This way asterisk can
provide the dialtone and the dialplan.
This also allows me to play a greeting to the phone
before
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Gabrielson
Sent: May 18, 2005 12:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am trying to get remote extensions to work correctly with
agents. I have ackcall=yes and have agents logged in to
extension 101 using agentcallbacklogin with extension 101 defined as:
exten = 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer))
This setup works great on local
On Sunday 15 May 2005 09:53 pm, Paul wrote:
Do you have the clout to get a handytone for evaluation and not have
salespeople calling you every day to ask how it's going? :)
Why not just buy one? You can buy one for less than $100 and if you
don't like it, you can just turn around and sell it
On Thursday 05 May 2005 10:27 pm, Tim Connolly wrote:
Pass through has the same functionality as a modem with a line and a
phone connection. Line is where you plug in the dialtone, the dial passes
through the phone connection unless the card picks up (like a modem
does).
I have a X100P clone
The Grandstream HandyTone 488 has an FXO port.
I've never used it though.
Cheers,
Jon.
On Thursday 05 May 2005 07:07 am, Chris Mason (Lists) wrote:
Is the Sipura 3000 the only way to interface a remote pstn line and connect
incoming calls to Asterisk? I have a location connected by network
The AG-168E has an FXO port?
The only seller I can find seems to think it is just a single FXS port.
http://www.iaxtalk.com/product_info.php?products_id=30
You wouldn't happen to have another link with more info would you?
Thanks,
Jon.
On Thursday 05 May 2005 01:33 pm, Joseph wrote:
On Thursday 05 May 2005 05:28 pm, Joseph wrote:
It has 1-FXS and one 1-Life Line (it is pass through type)
I've seen the pass-through term used alot and
I'm not quite for sure what that means. What is the
difference between a passthrough type and a regular
FXO. What can you do with one that
Is there a way to have an agent choose whether they want
to press # to accept a call on an individual basis when they log in?
Also, the faq mentions that you can play an optional message
to the agent before they press '#', how is this performed? The
queue message seems to play AFTER they
I've looked at both and can't seem to find mention of
either item in either place, could you possibly provide
a link.
Thanks,
Jon.
On Wednesday 04 May 2005 09:48 am, Matthew Boehm wrote:
Jon Gabrielson wrote:
Is there a way to have an agent choose whether they want
to press # to accept
On Monday 18 April 2005 11:23 pm, Joe Dennick wrote:
You probably can NOT use the existing Meridian phones because they are
digital phone sets, not standard analog ones. You can purchase 5
TDM400P cards (assuming you have 5 available PCI slots in your Asterisk
Server), and configure two with
I have 4 lines in a queue.
When a call comes in, I want all 4 lines to ring.
If all 4 lines ring and noone answers, I want to go straight to voicemail,
but if at least one line is busy, I want the caller to stay in the queue.
What is the easiest way to do something similiar to this?
Thanks,
That seems to only work on a single channel.
How do I use that for multiple channels at once?
Thanks,
Jon.
On Thursday 14 April 2005 01:57 pm, C F wrote:
I guess you could use the ${DIALSTATUS} variable
On 4/14/05, Jon Gabrielson [EMAIL PROTECTED] wrote:
I have 4 lines in a queue
And the stupid thing is that it is trivial to set up a script
to autorespond to these things. So assuming it is a
valid MX (which is easy to check for without harrassing
anyone), a spammer has an easier time responding
than a nonspammer.
Jon.
On Thursday 14 April 2005 06:14 pm, C F wrote:
Sorry to disappoint you, but questions only appear if the SELLER
wants them to. So if the seller doesn't like your question, he doesn't
have to make it show up on the listing.
Jon.
On Monday 11 April 2005 04:35 pm, dean collins wrote:
Lol, just posted a question to the list that should
You need a T1 card and a channel bank.
http://www.voip-info.org/wiki-Asterisk+Channel+Bank
Cheers,
Jon.
On Thursday 31 March 2005 07:53 am, David Hajek wrote:
Hi,
how to use Asterisk where I need to have lets say 40 analog lines. Any
ideas?
Thanks,
David
Ok, we all get it, some people prefer mysql, some people prefer postgres.
Now can we all just get on with our life or at least create a mailing list:
[EMAIL PROTECTED]
so that those people who think that mustard tastes better than ketchup have
somewhere more appropriate to argue.
Thanks,
On Tuesday 15 March 2005 06:34 pm, Robert Hajime Lanning wrote:
quote who=Giudice, Salvatore
So, let me see if I am right. You run a support shop? You want your
database to validate your data for you instead of leaving that logic
to
your application? Usually, a database is considered to
So are you saying that in my setup where I have a
adit 600 channel bank with FXO/FXS connected to a t110p,
that asterisk does an analog bridge? Presumably that
would mean 56k modems, etc.. would also work fine.
I was under the impression that asterisk used iax2 for the
internal trunk.
Jon.
On
You wouldn't happen to know how to do this would you?
I currently have a box with both hylafax and asterisk installed.
asterisk handles the dedicated voice lines over a t100p and
hylafax handles the dedicated fax lines over a 4port serial card
with external modems.
It would be really nice if I
I have some extra FXS ports on my channel bank that I could
plug the modems into, but is asterisk's fax support good
enough for a production system? Everything I seem to read
seems to state that asterisk fax detection and fax send/receive
support is still very unreliable or is this only over
A standard scsi cable works great.
Just cut it in half.
Cheers,
Jon.
On Tuesday 22 February 2005 07:17 am, Daniel Nyström wrote:
A little off-topic maybe, but it's still for the Adit used with Asterisk.
;)
I wonder where I can buy 50 pin Amphenol cables, with connector on one
side, and
How is the estimated hold time calculated?
Is this based on the average length of a call times number in queue?
Is this based on the average hold time times number in queue?
Is there some base number that is used before averages can be obtained?
Is there a way to set and/or tweak the estimated
I don't believe the adit 600 has an up/down for channels.
Are the channels connected to something. You might
look at the 'connect' command and see if that helps.
To bring the FXS channels up on my box I needed to
connect them to the T1 (in your case it would be the MGCP)
The t1 syntax is I
What are the advantages/disadvantages of using
a ZAP FXS port versus using one of the many
small ethernet FXS devices on the market. The
ZAP FXS talks directly to asterisk over PCI. Is this
an advantage? The ethernet devices I assume
speak either iax2 or sip, does this cripple the
Yes, this is basically the default.
Jon.
On Sunday 20 February 2005 02:20 am, Anton Krall wrote:
Guys..
Im new to asterisk but is it possible to simulate a dialtone for example,
in other PBX when you pick up the phone you can hear a certain dialup,
which is the PBX dialtone, and when you
I believe that is basically what the generic t100p is.
Also, several other voicemodems are already supported
by asterisk. To my knowledge they are all FXO ports.
I don't believe there are any modems that can provide
FXS ports. If someone knows of one, I would be interested
in knowing about it.
The feature that would be most useful to me is some sort
of rating feature on things like reliability, quality, etc...
where individual users could rate each provider. I'm not
for sure how best to handle something like this, but my
biggest problem currently is trying to decide which providers
Digium tech support recommends going with a t1 card and a
channel bank. This is by far the simplest, cheapest and cleanest
solution that I know of.
Jon.
On Friday 18 February 2005 09:21 am, Jim Van Meggelen wrote:
Folks,
In light of all the troubles people report when running more than one
I would think the best solution would be to have each
virtual PBX have it's own context and only run a single
instance of asterisk. This way you have less overhead
and can more easily manage the sharing of any
physical devices that you may have.
Jon.
On Tuesday 15 February 2005 12:07 pm,
You didn't say what your fxs/fxo requirements are but:
A T1 card ($500) and a used channel bank ($300) might be
a good alternative.
You also might check out the voicetronix cards.
Cheers,
Jon.
On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote:
Are there any other relatively low cost
You need to tell us what type of device you are
using to make the phone calls. Are you using
a ZAP FXS, a softphone, a sip phone, or an iax phone.
Also, how are you terminating the call. Is it via a
ZAP FXO device like a t100p, is it another VOIP phone,
or is it via a service provider like
Are there any gui's that support zap fxs extensions?
AMP seems to be one of the more popular gui's but
it doesn't support zap fxs devices.
Thanks,
Jon.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
They also have a .tar.gz file if you don't want to use the iso
and my understanding is that the .tar.gz file does not nuke
your harddrive.
Jon.
On Wednesday 09 February 2005 11:04 am, Brett, Gary wrote:
Thanks Dean, you say that [EMAIL PROTECTED] is just an automated way of
installing AMP
Some things that I might check:
1) make sure you have set the jumper to E1 or T1
Mine came configured for E1 and I needed T1.
2) try to see if you can get it working with a different
configuration.
i.e. My configuration for a wcte11xp configured for a
T1 channel bank is:
loadzone = us
I started a small bounty for it at:
http://www.voip-info.org/wiki-Dialtone+and+line-in-use+detection+on+a+ZAP+channel
Hopefully other people will add to it and make it grow.
Jon.
On Monday 31 January 2005 12:02 am, Paradise Dove wrote:
On Sun, 30 Jan 2005 18:14:38 -0600, Jon Gabrielson
The digium X100P is probably fine. What most of these people
are talking about are the X100P clones which are of varying
qualities. I have had no problems, but results vary.
Cheers,
Jon.
On Sunday 30 January 2005 09:42 am, dean collins wrote:
Why does the X100P have echo and the Wildcard
Can't asterisk look for a dialtone? Even a $5 modem
can detect whether or not there is a dialtone.
Thanks,
Jon.
On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote:
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already
, Steven Critchfield wrote:
On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote:
Can't asterisk look for a dialtone? Even a $5 modem
can detect whether or not there is a dialtone.
Maybe you should just use your $5 modem and write your own software.
Asterisk is a PBX. PBXs shouldn't have
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten = _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
in the background.
I have asterisk connected to an adit 600 with fxo ports.
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
used by someone else. Any ideas on what I'm doing
wrong?
Thanks,
Jon.
.
On Saturday 29 January 2005 11:11 am, Steven Critchfield wrote:
On Sat, 2005-01-29 at 10:36 -0600, Jon Gabrielson wrote:
I have asterisk connected to an adit 600 with fxo ports.
When I place a call with asterisk, asterisk will try to dial
out on the first line even if the first line is already being
From the asterisk demo:
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,1,Answer ; Answer the line
You can also wait 10 sec, 30 sec, etc... to allow as many rings
as you like.
Cheers,
Jon.
On Friday 28 January 2005 09:08 am, Steven P.
I have an adit 600 with an fxo card connected to a digium T1 card.
If I try to make an outgoing call and the T1 cable is disconnected,
asterisks returns congested like it should.
But, if the adit 600 is connected to the T1 card, the adit 600
immediately answers the call even if there are no
On Friday 28 January 2005 05:41 pm, Steven Critchfield wrote:
On Fri, 2005-01-28 at 11:35 -0600, Jon Gabrielson wrote:
I have an adit 600 with an fxo card connected to a digium T1 card.
If I try to make an outgoing call and the T1 cable is disconnected,
asterisks returns congested like
assuming a ZAP interface, just set
callwaiting=no
callwaitingcallerid=no
in zapata.conf
Cheers,
Jon.
On Friday 28 January 2005 06:00 pm, Alex Barnes wrote:
Hi all,
Hopefully you can help me.
I want to turn off the audio Call Waiting beep that plays during a call.
I have found the
I have an adit 600 connected to a normal analog line. When I try
to call that line, the phone rings a quarter ring(almost a beep) instead
of a complete ring and keeps ringing and ringing with asterisk never
picking up the call. Outgoing calls on those same lines aren't working
either.
Any
or normal
phone and attached it directly to the outside line to see if you get
dial tone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jon Gabrielson
Sent: Friday, January 28, 2005 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial
I just bought an adit 600, and it works great.
They can be picked up used pretty reasonably (approx. $10-$15/port)
It also does callerid and callwaiting great, but seems to have
problems doing callwaiting callerid, as do a lot of channel
banks I believe, so if this is something important to you,
Does anyone have any experience with iax.cc/sixtel?
Are they a legitimate company? From their website
it looks like you can get a private incoming 800
number for 30 cents/month plus 2 cents/minute.
Somehow that pricing seems a little cheap for a
DID number. I assume there has to be some
We switched from postgres to mysql several years ago
for several reasons, one reason being that mysql is
considerably faster. The other reasons had to do with
certain functions that postgres didn't support, I have no
idea if they now support them or not, but we have been
very happy with the
What is the best way to display queue status to a station?
Our current phone system has 4 lines at each station so
it is really easy to see how many lines are waiting. I could
replicate this using a 4 line phone, but this requires both
running an extra 4pair to each desk as well as taking up
4
On Monday 24 January 2005 11:18 am, Manjit Riat wrote:
Hi,
I am thinking of signing up with voice pulse connect to connect to my
asterisk server and using it as a regular line. Is it good? Or should I go
with vonage or others ?
Vonage doesn't allow unofficial devices like asterisk.
See
Does anyone have any experience with making an
adsi phone appear to have more than one line.
It seems like this would be a very simple and very useful
thing to be able to do. Ideally, it would be nice if you
could make the 6 soft buttons appear as lines 1-6 and
if you press one of the soft
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