figuration, as well as managing updates. Trixbox now uses tidy,
simple binary rpms for updates so this will go along ways to ensuring
stable/simple/quick upgrades.
Thanks for any suggestions
:)
-----
Johnny Stork Open Enterprise Solutions
http://www.openent
I have something called an EnterPhone 2000 intercom system in my complex
which rings the phone when someone dials my buzzer number on the keypad. I can
use any Asterisk extension to anser the call and hit 6 to open the door.
However, I have tried using a Digitial Assistant with the message If
I have a fairly new, but functional install of [EMAIL PROTECTED] 2.7 with a
TDM400 (1 FXS) and T101P (1 FXO) hardware. For some reason the analog phone
connected to the FXS port and one SIP softphone goes straight to the voicemail
indicating Is On the Phone although it is NOT off the hook.
I seem to be having a problem with my GXP-2000. No matter how carefully I type
in the mailbox number and password when calling the mailbox (*98), it keeps
complaining that the password is not correct? I can use any other phone to
check the same mailbox and it works fine, just not with the
7:48:30 DEBUG[4600] chan_sip.c: update_call_counter(103) - decrement call
limit counterApr 28 07:50:11 DEBUG[4485] manager.c: Manager received
command 'Command'Apr 28 07:50:11 DEBUG[4485] manager.c: Manager
received command 'Command'
-Original Message-From: Marco Mouta
[m
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
Make sure you have the DTMF mode set to RFC.
On Fri, 2006-04-28 at 15:20, Johnny Stork wrote:
I seem to be having a problem with my GXP-2000. No matter
how carefully I type in the mailbox number and password when
calling the mailbox (*98
Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not
seem to be a menu area/settings for Incoming Calls?
If you have a similiar setup, or know what the settings should be, could you
possibly post them? If I were to create a dial group
to ring all extensions, could that
)
-Original Message-
From: Johnny Stork
Sent: Thursday, April 27, 2006 7:11 AM
To: asterisk-users
Subject: RE: [Asterisk-Users] Unable to accept incoming PSTN calls
Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does
not seem to be a menu area/settings for Incoming
route with blank
CID and DID, and point it where you want it to go. It should work after that.
Alex
On 4/27/06, Johnny
Stork [EMAIL PROTECTED]
wrote:
Since
I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu
area/settings for "Inc
With so many vendors offering so many different bundles, packages and
various PBX systems, can anyone suggest either/or a well know, reliable
SOHO sized VoIP/PBX, based on Asterisk, as a software-only or
appliance bundle?
Basically I am looking for something economical (under $500.00), which
I
I am new to Asterisk and the protocol/language complex world of VoIp and PBX.
But I have a dedicated machine running [EMAIL PROTECTED] 2.8, a single TDM400P
with one FXS module card connected to a standard analog phone. The second card
is an X100P connected to my analog PSTN phone line. I also
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