> Steven Critchfield wrote:
>
> > On Fri, 2003-09-05 at 03:25, Marcel Prisi wrote:
> >>This is a completely open-source and open-hardware hardware phone
based
> >>on Linux on an ARM embedded platform ... they already had lots of
> >>experience ... but might need some different software ...
> >
>
BTXML support for client applications is necessary to achieve this. The
SIP images state that they support BTXML; however, they only use it for
their internal screens and internal navigation. CMXML is the only
language supported for client applications with the SIP loads currently.
A little bird at
Isamar,
> I need to make a little IVR app and get/send the data
> into a MS-SQL database.
> As far as I know, it doesn't have driver for Linux.
> Anybody here already found here any workaround for this situation?
> Maybe, I can use an AGI interface to do that, maybe perl+ODBC?
There is no support
> > I feel like a T1 with 24 channels should suffice, but what exactly
do
> > I order and what to I have to have in my asterisk unit to interface?
> > Does the line they terminate just plug into a T100P or do I need
some
> > extra hardware? What services do I need to be sure I order on the
T1?
> >
OK, this is probably a dumb question for a lot of you, but I have no
experience with digital lines outside of a tiny bit of ISDN, so I'll
just bite the bullet and ask some newbie questions. I am attempting to
plan an asterisk installation with about 20 SIP phones and the following
incoming lines:
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian Capouch
> Sent: Thursday, July 24, 2003 1:02 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Changes to reset method for ATA186?
>
> I am trying to do a "factory reset" of a
In the US, and probably some other English speaking countries, # is the
"pound" key on telephones. In the UK it's called the "hash" key. The
technical name for the punctuation mark is "octothorpe".
A lot of punctuation has strange technical names that you don't hear
every day (or maybe you do if
> I've seen references to this module in the mailing list archives, but
it
> isn't in the 0.4.0 tarball, nor is it in CVS. I can roll my own and
was
> planning to do so anyhow, but that doesn't seem to make a lot of sense
> if it already exists. Am I not looking somewhere I should be looking?
Most
> Who else on here prefers the newsgroup/threaded approach? If you
haven't
> already, check out news.gmane.org for mailing lists turned into
newsgroups
> readable by news readers...
If you want threads, get a MUA that is capable of threading. Most are.
The In-Reply-To header makes mail thread
I have written a small perl CGI script that demonstrates how one might
use the asterisk spooler 'pbx_spool' to make a 'click-to-dial' type
application. The script is intended to be a demonstration example only
and since it has little security, should not be deployed. I was just
experimenting with t
> Other things:
> The phone constantly says "Ethernet Disconnected". Even though it
tftp's
> configs and registers with the proxy.
Something is wrong with either the phone hardware itself, the network
port on the hub or switch it is attached to, the ethernet cable it is
connected with, or the 10/1
> I'd like to have a SIP phone at home and at the office and have them
both
> ring when my extension is dialed. Right now I used the same config for
the
> phones (Cisco 7960's). So they both register with the same login & pw.
> This
> doesn't seem to work quiet right, where only the last phone to
r
> Also almost forgot. They sell the demo voices on their site for 29.99.
> Linux and windows versions. Since I believe what they use is based off
> festival, perhaps the voices could be made to plug into the existing
> festival plugin for asterisk?
I have been working with app_festival for about a
> Hey all,
>
> quick question: does asterisk work okay in a Cygwin environment?
>
> I want to install it on my cygwin setup for local testing/demoing and
save
> me the hassle of using a pure linux machine
I had suggested to the fellow asking about running Asterisk in VMWare
(It won't work
> Thanks for the heads-up. Do you know of any alternatives?
There is the zaptel hardware from digium.. the TDM400P for FXS ports,
X100P for FXO, or the combination of T400P + channel banks.
> I recently posted a similair query to comp.linux.hardware relating to
> setting up a virtual telephone e
Dan,
Your problems are all the result of your computer and your software.
It's not going to work for you in your setup. Repeat: It's not going to
work for you in your setup. Repeat again for increased clarity: It's not
going to work for you in your setup. I really don't understand why you
keep ask
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk
> Sent: Monday, July 14, 2003 3:16 AM
> To: [EMAIL PROTECTED]; cvasiliu
> Subject: Re: [Asterisk-Users] module : cdr_sybase.so
>
> nice
> this can probably be used w
> You can build a UPS for that, but the better option here is to attach
a
> phone to the phone side of the X100P that is always connected to the
> POTS line so that even when the computer goes down you can send and
> receive calls.
If you don't want it to ring *unless* the power is out, you could
I noticed that the BudgeTone (I have the 102) with the latest firmware
tries to download a file called cfg.txt (presumably the configuration)
and a file called ring.bin (presumably a ringer) from the tftp server.
The ring-in sound on the budgetone is the same as the ring-out sound and
that is goin
> I use a similar asterisk setup when my T100P card is being used
outside
> the home. I ran it on a 1ghz athlon that was shared as my workstation,
> but all linux. When my screen saver started, the audio quality dropped
> below usable. So obviously my system was idle if the screen saver
> started.
h connection. Plus, the problem also happens when
reading festival speech from the cache that app_festival can create, and
that action most definitely does not need to be serialized.
Thanks anyone, for any insight you can offer... I am beginning to
understand the * i
> Does anyone know if someone makes a hard video phone for SIP.
>
> Dave
I was curious about this too as the video support has been going in.
Below are links to what I found. As with some of this stuff, I can't
really find who the manufacturer is
Leadtek is possibly the manufacturer of this d
For anyone who is interested, I have a working tftpd (modified wvtftpd)
capable of serving configuration, dialplans, and ringtones to Cisco
7960/7940 and ATA-186 devices that are located behind NAT firewalls. As
TFTP is not a very firewall/NAT friendly protocol, I had to break some
rules to get it
> In my experiences with Cisco 79xx phones, if you have
> "proxy_register: 1" set, then the phone will require a successful
> REGISTER transaction before it will give you dialtone. Try setting
> that option to "0" and see what you get and if it works with
> "defaultip=" below.
The phone is runnin
I have several (various brand) sip devices with static IP's.
I understand that asterisk will not accept a registration from these
devices if the host= parameter is not set to 'dynamic' in sip.conf.
I want calls to these extensions to be routable even before the device
registers. I understand that
Would it be feasible to look into implementing a way to preserve SIP
(and possibly other protocols') registrations during a scheduled or
otherwise requested restart of asterisk? I am thinking somewhat along
the lines that the SIP registration has a certain duration during which
it is valid that is
Cool trick!
You could simplify this:
[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten => s/_,1,Dial(${ARG1}r2,${ARG2})
; Caller*ID is not 4 digits (external call)
exten => s,1,Dial(${ARG1},${ARG2})
; Both of the above lines go here next
exten => s,2,Voicemail(u${MACRO_EXTEN})
> So far, I've only been able to get the XTEN Lite phone working
> and I really don't understand how I set it up. I used "xten"
> for every option everywhere (display name, username, password,
> and Domain/Realm) and the corresponding section in sip.conf.
> I've had no luck getting the SJ Labs soft
e number like this:
>
> exten => _X.,1,Dial,Zap/1/w${NUMBER}
>
> You could also try to put 'w' inbetween the digits.
>
> regards
> Martin
>
> On Tue, 17 Jun 2003, John Laur wrote:
>
> >
> >
> > Quite frequently, outgoing calls fr
Quite frequently, outgoing calls from the X100P cards here will not dial
properly. Instead of hearing the ringing after the Zap interface picks
up, I'll hear silence for a while then the 'If you'd like to make a call
please hang up and try again' recording as if zaptel picked up the line,
punched
> > Did you cvs update zaptel and recompiled ?
>
> Yes. I followed the instructions on the Digium download page, namely:
I looked at the log file and there was no commit on this recently. It
seems that if this change has been made, it's just not in CVS yet :)
Looking forward to trying it though..
> After carried out some local tests, the BT engineer confirmed that the
> detected line fault was caused by the X100P. We have disconnected the
> X100P from the PSTN line for now in case it would trigger the line
> fault test again (and we'll be charged for the engineer visit). I
> suspect that th
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