Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
interface with Skype and Asterisk. Basically, make a Skype ATA, but
1 FXO per PSTN, so you would need 16 FXO ports. That would be accomplished
by 4 TDM100P with 4 FXO modules on each.
Undrhil
--- Asterisk Users
Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com
wrote:
If I have 16 PSTN for my trunklines, how many FXO do I need?
Thanks.
I asked about a similar application a few weeks back. This is sometimes
referred
to as campusing since you are basically going to make the two systems sharing
their resources appear to be one system. From what I understand, you have
to have both boxes running Asterisk. I am pretty sure that
I thought Background() only allowed you one digit dialing while it's playing.
Is this not the case? I agree with the reply which said that you want to
use DISA, the only problem with DISA is that you have no way to use the line
for answering regular calls. Once you put the DISA command in the
I just realized that I came up with a way to use DISA and still allow for
voicemail-like activity. Set background() to play your greeting and then
program in an exten with 9 which would then point you to your DISA dialtone.
If the caller doesn't press 9, have them go to the general leaving a
Hey everyone,
I recently bought an Act-Tel G2DS telephony gateway (the
web interface says it's model # is GS though.) Has anyone else on this
list used one of these? It has one FXO and one FXS port. I have an account
for it set up in sip.conf on my Asterisk box and it apparently logs
That was the first thing I checked for. I have since discovered that by
checking
off the comfort noise selection in the options for the SIP account and then
restarted kiax, I don't get the noise anymore. I guess it's something that
kiax uses as a way to make sure the caller knows I'm still on
OK, here's another problem I've run into with Asterisk. In the musiconhold.conf
file, I can set the music on hold mode to files and the directory to the place
where I have my MP3s stored and they play. If I set the music on hold mode
to any other setting, instead of getting my MP3s, I get
Call Waiting will not allow you to simulate rollover because the second incoming
call will simply beep on the current call on that POTS line. As far as I
know, Asterisk has no way of picking that call up independently of the first
call.
Undrhil
--- Asterisk Users Mailing List - Non-Commercial
Does SayDigits(${EXTEN}) not work in this case? I would imagine that it would
still maintain the dialled extension in that variable, would it not?
Undrhil
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
wrote:
I am trying to modify a fairly complex
--- Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
wrote:
On Jun 13, 2006, at 7:52 PM, Josué ƒonti wrote:
?? Doug,
If you it will not have hardware and if ztdummy will not have
installed
its moh will not function correctly
I believe this is no
Has anyone on here used kiax before? I am asking because I have it installed
on several computers and have been able to get it to connect and register
to my Asterisk box. I can even call between them and my SIP softphones.
The problem I am having is this: when I use kiax to call someone else,
Hey. Maybe you can give me a hand with configuring my Linux box to send out
emails? I've installed sendmail as per *several websites* and it's installed
and running. I've gone into the voicemail.conf file and specified to allow
attachments, etc. And, yes, I restarted Asterisk. Technically, I
Basically, I am looking to set up an extension which will be used as a
help-line.
I want it to function kind of like the bat phone from the old Batman series,
where Commissioner Gordon would pick up the extension in his office and it
would ring the phone back at Wayne's mansion. Is there a way
Easy to do on the Linksys PAP2, if that helps. The functionality
probably
depends on the make and model of the phone... maybe if you gave
those details
as well?
James
Well, there's the rub. I don't have any of the
hardware yet. I am looking at the various options before buying anything.
As silly as it sounds, I found the demo setup which was shown in the video
by Systm (www.systm.org - episode 5) works really well for testing these.
:)
Once you get the SIP phone configured in sip.conf and you get the phone
connected, you can make the following context in extensions.conf to
If the end-user PCs are setup with speaker/microphones or are using headsets
with boom mikes, you could look around at some of the free softphones, like
X-Lite or their pay-for cousins.
Unfortunately, phones of any quality higher
than 1-line, caller id, speaker phone are going to cost some money,
So, your dialplan for that incoming call is just the one line?
exten =
s,1,Dial(IAX2/carey)
Nothing else? Try adding a Hangup command on the
next priority and see if that helps any.
exten = s,2,Hangup
If you
already have a Hangup command in there, then I apologize for wasting your
time. :)
I want to set up a test system at my house using a FXO card (one of the X100P
cards) and a FXS device. I am going to have the one line running into the
machine and the house phones running off of the FXS port. I will be installing
a softphone on my laptop and will be using Hamachi to tunnel back
I have been looking around some and I can't seem to find anything which will
answer my question. If I have two Asterisk boxes in different locations which
are linked to each other over the internet, can I configure the boxes to use
each other's lines as local?
In other words, let's say Site A
Hello everyone.
I had heard about this open-source PBX once a while back.
I wasn't too interested in it at the time but I kept the info filed away
for possible future use. A couple of days ago, I was walking around Barnes
and Nobles and I found this book, called Asterisk: The Future of
For Problem #1:
exten = _X.,1,SetGroup(${EXTEN})
exten = _X.,2,GotoIf($[${GROUPCOUNT}
= 1]?104:3)
exten = _X.,3,Dial,SIP/username
exten = _X.,104,voicemail(u${EXTEN})
exten = _X.,105,hangup
This will limit the amount of incoming calls
to 1 and send everything else
to the VM.
Hey. I was
Yes you are correct... by default asterisk will send the call to priority
N+101... what is your point?
You asked about turning off call waiting.
In the example that I provided,
if the amount of active calls is 1 then
it will forward to VM without
dialing the exten. That is what you asked
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