Is there a phone in the sub-400$ department that has at least a 10 inch display
and a nearly modern web browser with keyboard hook-up that is either sip or iax
based? I dont want to do video, just have a good web browser built into the
phone.
___
--Ba
Its for a collections type application. Accounts passed due will show up on the
screen in either an automated call fashion, or as a result of an incoming call.
I need the physical interface to be somewhat simple since the end users are not
exactly computer friendly. I had considered using the po
I am looking for hardware recommendations for a new asterisk setup. I want to
connect 8 analog phone lines (I think the term used is POTS?), to an asterisk
server similar to the way the Wildcard TDM400 with FXO cards would. However,
The Asterisk server is rack-mount lacking the needed PCI ports,
On Jun 30, 2005, at 14:53, Chris Gamble wrote:
> I am looking for hardware recommendations for a new asterisk setup.
> I want to connect 8 analog phone lines (I think the term used is
> POTS?), to an asterisk server similar to the way the Wildcard
> TDM400 with FXO cards would.
Does this just sound worse than it is?
>> "With SBC you are out of luck, since Asterisk doesn't detect dialtone (
>> it dials blind, sometimes too quickly for the CO to catch the first
>> digit, resulting in wrong numbers )) or stutter dialtone either, and
>> reportedly has had any indication
How adventurous would a person have to be to try to use the 1.1 from cvs? I
want to implement our phone system with the database connections built in,
which as I understand is being made very easy in the 1.1 code that is under
development.
thanks,
___
I am looking at phones for my asterisk system and seem to have a problem.
The only Power over Ethernet phones I can find that support the IEEE
standard are 3com. Cisco uses its own proprietary ( and is expensive to
boot ), snom has a different but equally non-IEEE method, and i'm havent
found anoth
>From their website, the key difference between the polycom 500 and 600 phones
>is the number of "lines" they support. What does this mean in terms of
>asterisk? Do I have to have a seperate extension for each of these lines or ?
Also, slightly off-topic, how does the 500 POE "optional" cable w
I am using the July 1 release of Asterisk @ home for an initial test-bed. The
one and only thing my users have complained about so far is that the directory
spells out the name of the person. Is there a way to get it to "say" the full
name of the person?
Thanks,
I have seen various other problems with the cards not detected, but I seem to
suffer a different fate. The system boots and recognizes fine, but when I call
in from an external line, all I hear is a horrible static. The system works
fine when only 1 card is present. I have already moved the new
Interesting story. I was following up any wild goose chase i could to find
a resolution including cleaning dust out of the machine. At 4 pm I plugged
the line in and called for the sake of luck, and it was back to working.
In the mean time, thanks for the troubleshooting advice. I will keep this
i
I have done research and reading around text to speech, and was wanting to
get an updated query of where everyone is with this. I have installed
festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home
ISO ). I also changed the directory php application to use the festival.pl
to rea
I just received my first polycom 501, tried my best to follow all of the
documentation and configs on the wiki ( though some conflicted in which
case i tried both ways ), and at the end of the day can not connect my
phone to asterisk.
My Questsions: does the wiki information apply to the 501's? Th
If the RX has a constant 2 # marks when no one is on the phone, thats a bad
phone line, right?
<(RX)> <(TX)>
##*
Rx: 493 ( 496) Tx: 0 (0)
___
Just got in a bunch of polycom phones for use on my shiny new asterisk box, but
found 2 small issues I was wandering if someone could help me with.
First, though the phones support 2 call appearances, if I am on a call, the
second call does not ring through -- it goes to voicemail instead of let
>> Just got in a bunch of polycom phones for use on my shiny new
>> asterisk box, but found 2 small issues I was wandering if someone
>> could help me with.
>
> Are you using AMP or Asterisk @ Home?
>
>> First, though the phones support 2 call appearances, if I am on a
>> call, the second call does
I have 2 TDM04b cards currently running in an asterisk at home box that I am
ready to replace with the CVS version of asterisk. What I am looking for is
thoughts / recommendations. I want to move this to a small form factor (
shuttle ) machine and was wandering what expeience / advice there was
Chris Gamble wrote:
>>I have 2 TDM04b cards currently running in an asterisk at home box that I am
>>ready to replace with the CVS version of asterisk. What I am looking for is
>>>thoughts / recommendations. I want to move this to a small form factor (
>>shuttle
I have seen in several places where the analog adapters you can use for faxes
and modems have intermittent problems. Is this an issue with asterisk and are
there work-arounds?
We currently have side-stepped the issue by sharing the analog line between the
tdm card and the fax machine (I assume
I have run my soundstation out of the soundpoint directory, and now get a
repeated error message of: Application sip.ld is not compatible with phone. I
found the warning AFTER turning the phone on within the network. Can this be
fixed?
___
--Bandwidth
We have occasional echo, mostly on lines that come across as quiet.. I am
trying to troubleshoot how to disolve the echo. i have tried the steps on
voip wiki, but have not recently been able to make any difference.
the echo isnt horrible most of the time, and seems extremely random in
that i can c
We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are
getting frequent restarts on the spans which lead to dropped calls. I have
pasted some hopefully pertinent information below -- anyone have any clues that
might help?
Thanks
Next line is repeated throughout messages,
ling List - Non-Commercial Discussion"
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
----- Original Message -
From: "Chris Gamble" <[EMAIL PROTECTED]>
To:
Sent: M
23 matches
Mail list logo