the best way to go about this is,
should I just forward existing lines to specific phones (just to save on
running new telephone cabling) or would there be any simple ways to make
a small menu and just put one more layer before they get through?
Thanks
Joel Duffield
Near North Business Machines
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote:
> We want to use asterisk to extend our current phone system. It is a
> regular plain old system. Has anyone done this before?
Absolutely - in a lot of different ways.
> We would be
> adding about 4 SIP (probably Cisco) phones
question is, does my logic work, and also if I use the dial command,
and I set the analog system to pick up immediately, will wait long
enough before it dials? If that wouldn't work is there a way that I can
tell * to dial then wait and then send digits?
Thanks
Joel Duffield
Near North Bus
price range ($300-400) that can handle 4 Fxo's.
Thanks
Joel Duffield
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w can I get asterisk to pick up a line and send these
digits
after a voicemail has been left, even if the person hangs up the phone? would I
need a script to do this, I'm a noobie at writing scripts? Any experience or
advice
would be greatly appreciate
has anyone used the Clipcomm 4 port FXO Gateway with asterisk? and has it
worked just as a gateway should and pass calls straight to asterisk, and be
very easy to place outgoing calls on? This is one of the cheapest gateways I
have seen and I need to have the ports out of my asterisk box so that I
When I put a caller on hold who is connected via our IAX provider the MOH is
dreadfull, not acceptable at all. it come on for one second then off for a
few. The calls sound fine when I am taking to someone, and the MOH works on
the network, I also have a dummy timer installed. Has anyone had this
p
ï
This
file is used for the Directory Application. I don't think it is ever used in
voicemail, it's only used to play the name before Directory forwards them to
that extension.
Joel
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
bamSent: T
I would think what you would need to look at is how to do this with the *
Data Base. I haven't done this, but it would seem that there is a way to
make it work with that.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert P.
McKenzie
Sent: Saturday,
Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
can
e beeping?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?
Ok my first question is I have seen messages abou
Hi All
I had asterisk running on a xercom install, I upgraded the box to a full
debian install and now asterisk is not starting from on boot. I can start
asterisk from the command line fine no problems, but when i type
/etc/init.d/asterisk start it says asterisk PBX started. It doesn't start it
th
I have read how to get paging working on the 841 using the SIP Header. I
have tried to install chan_sip2 but the make failed and the patch that was
also mentioned I cannot find to download. I am using asterisk stable 1.0.7.
What is the best way to implement this with the littlest cost to other
feat
I have an * 1.0.7 box that keeps loosing its registration to both of the
other servers it registers with. When I start * it connects and registers
fine and I can make calls. but after a few hours it shows the status as
Auth. Sent but I can no longer make calls to the other servers. The other
server
My * box keeps losing its registration to all the servers it is registering
to, the only way to fix it is to restart asterisk and then it works fine for
another 2 hours or so. I'm on a static IP, but this happens like clockwork
every time. I have seen other people that have this problem but never a
Hey steve
I remember a tip somewhere where they used a conference room and added all
the users into that conference muted, then kicked them out at the end of the
call. Sorry I can't remember at all where this was but it looked like it
could work. How did you get the autoanswer to work, I have trie
Try to use macro's I am not the one to ask about them, I couldn't give you
an example off the top of my head. But read up on them on the wiki, and i'm
sure they can do what you want very easily.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
St
Okay sounds like a stupid question but just to be clear do you have some
sort of timer on both machines?
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed
Sent: Saturday, May 21, 2005 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial D
005 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX losing registration
Maybe you connections pass through a stateful firewall , and these states
die after some inactivity time... Check it.
- Original Message -----
From: "Jo
ssage-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Riddell
Sent: Saturday, May 21, 2005 11:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX losing registration
Joel Duffield wrote:
> The firewall I'm using is a Lin
Discussion
Subject: Re: [Asterisk-Users] IAX losing registration
Joel Duffield wrote:
> The problem is still occuring. it happens even if I register with myself,
it
> works for some time and then just dies. The qualify still shows up as 65ms
> on the outside server, but the registry just says
Has anyone had problems with a small electrical type of hum on the 841's
handset. It is there on all of the three phones I bought, and also do the
sound like the microphone is cheap and kind of a high pitched talking into a
can. I can live with these as long as I know that this is what the phones
a
I am using asterisk as a voicemail for an old tie onyx phone system. I have
almost everything working except that the stupid phone system doesn't hang
uo when the user hangs it it only sends me a bunch of 9's. I put a 9
extension that hangs up and that works for the Background app, but once I
get i
I am using asterisk as a voicemail for an old Nitsuko TIE ONYX VS system. it
sends the signals through DTMF, and that works ok sometimes. but when a call
comes in from outside and is transfered to voicemail or goes to voicemail
because no one picks up, the following happens.
the interface rings
as
WAV: No such file or
directory
Does anyone know what is up with this apache isn't trying to share this
anymore so did it change some permissions?
Thanks
Joel Duffield
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