You can have two calls per line appearance. If you assign both line
appearances (both can be the same extension) you are allowed four calls.
On Mon, 23 Aug 2004, Christopher L. Wade wrote:
Hi all,
I know this is a stupid question, but it is one I've been trying answer
for quite some time.
On Thu, 19 Aug 2004, Peter Harrison wrote:
I've been tasked with fitting out an appartment building with a phone system.
The entire appartment is getting broadband, with Cat6 Cables into each
appartment and fibre down the trunk. I was hoping to have PSTN in the
basement connecting to the IP
On Fri, 30 Jul 2004, Darren Bentley wrote:
Hello,
Has anyone used Asterisk in conjunction with a billing system like
Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used?
I suffered with Rodopi for three years in a previous life. Avoid it like
the plague.
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten =
On Sat, 31 Jul 2004, Kevin wrote:
Does anyone know if the 480i supports 802.1Q?
I don't see any support for it at the moment, but this is a very early
firmware, with a bare minimum of features.
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On Sun, 1 Aug 2004, Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and still be able to hear the parking lot stall number.
Unfortunately, I have no idea where
I have an * server at my office, lighting message lamps on SIP phones
fine. I also keep an * server at home, with no Zaptel hardware, and hang a
SIP phone off of it so I have an extension at home with no SIP/NAT
ugliness. Is there a way to light the message lamp at home when I have
voicemail at my
For those that are interested, here is my report back to Sayson on the
480i
-- Forwarded message --
Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT)
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: 480i User Feedback With Asterisk
Seshu,
I am using a 480i, and I am impressed
On Fri, 23 Jul 2004, Brent Franks wrote:
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is
On Mon, 19 Jul 2004, Ryan Courtnage wrote:
Hi all,
We find that our UIP200s clip off the 1st second of audio prompts from * (ie:
the beginning of voicemail prompts).
Has anyone found a way around this?
Running CVS-D2004.06.29.15.30.00 on WBEL 3.0
This happens with my 7940s as well. I
On Mon, 19 Jul 2004, Ryan Courtnage wrote:
This happens with my 7940s as well. I have found that using and Answer,
and a Wait(1) before playing back prompts works well. Prevents Alisson
from saying Assword? when dialing VoicemailMail(20).
Thanks for your reply. I have been able to use
On Sun, 18 Jul 2004, Michael Welter wrote:
Does anyone have a recommendation for a 48 port LAN switch for a new *
system? I'm not happy with NetGear's reliability.
You can get Cisco 2950s for about $600/24 ports.
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You need to send a vallid CALLERID to Nufone.
On Sat, 10 Jul 2004, V59Net wrote:
Use the NuFone to call numbers 1800,1866 and after 20 seconds the call is
interrupted. In log of * it writes: Max retries exceeded you host.
Somebody can help me?
___
On Sat, 10 Jul 2004, Brian K. West wrote:
No you don't it will just make one up...
I beg to differ, Mr Brian sir. I had problems calling 800 numbers with
Nufone, and Jeremy explained to me that they check for a caller id before
sending calls to 800 numbers.
On Fri, 9 Jul 2004, Antti Lohikoski wrote:
and No identd (auth) response followed with Closing Link: StiX
(Invalid username [~antti.loh])
Maybe your username is invalid.
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On Fri, 9 Jul 2004, Andrew Kohlsmith wrote:
On Friday 09 July 2004 06:54, Antti Lohikoski wrote:
1. The irc.freenode.net server gives me Couldn't look up your hostname
and No identd (auth) response followed with Closing Link: StiX
(Invalid username [~antti.loh])
This is *specifically*
On Wed, 7 Jul 2004, Ryan Courtnage wrote:
Hello,
Over the past several weeks, we have been having an intermittant problem with
our deployment of a TDM400P card (4 fxo). We have tried many things, and the
problem still re-occurs.
The Problem:
Occasionally (every 48 hours), the TDM400p
On Wed, 30 Jun 2004, Brian McSpadden wrote:
What settings are you using for rxgain/txgain? Do you have echotraining=yes on?
On Mon, 28 Jun 2004 00:45:52 -0500, Chris Foster
[EMAIL PROTECTED] wrote:
I Just finished getting HEAD running (had to update to Slackware 9.1
from 9.0 to do it)
On Tue, 29 Jun 2004, rich allen wrote:
iH
using Slackware 9.1
after install the card (new) i get the following from dmesg
Module 0: Not installed
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (FCC mode)
how can i determine if this is a hardware
On Sun, 27 Jun 2004, taf taffey wrote:
Hi,
I've tried the latest CVS Zaptel and Asterisk after following the Echo fix threads.
But echo is the same if not worst.
Has anyone managed to alleviate their echo from these latest changes?
My echo has vanished entirely with the latest CVS.
On Tue, 8 Jun 2004, John Campbell wrote:
Hi,
Having searched through the mailing list archives and the wiki, I still
don't know how to solve the following problem:
Call is received, phone rings once, then the caller gets the voice menu.
What I want is for the call not to actually ring,
On Mon, 7 Jun 2004, Kurt wrote:
The Cisco 7960 has a softkey called DND which when
pressed as the phone is ringing will sack the call to
voicemail. If you where using Cisco CME or CM you
can forward all calls to Vmail via CLI or GUI.
It does? I have 7940s, and the DND is buried deep inside
I am trying to run am-web on my asterisk server. The machine is CVS-HEAD
from 5-29-2004, on Debian Testing, running Apache as httpd.
If I untar the am-web.tar.gz file to /var/www/am-web, and access
http://office.bgcfreedom.com/am-web//command.php?page=listsip or any other
command in a browser, it
I see the same thing:
marconi*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
192.168.1.100(None) 000f9054-3a 00101/02439 UNKN
192.168.1.103(None) 000f9057-96 00101/01079 UNKN
192.168.1.101(None) 000f9048-5d 00101/00113 UNKN
On Wed, 26 May 2004, Rich Adamson wrote:
Has anyone tried the current Head cvs with TDM04b (4-port fxo)?
The card stopped answering inbound calls (no CLI indications whatsoever),
although outbound pstn calls via the card work just fine.
Kind of looks like one of the changes from yesterday
Welcome to Voicepulse and their lack of jitter buffer. This is the
cause of your horrible sound. Will be just as bad with SIP.
Which providers give you a jitter buffer?
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Any clues as to what this is? Google isn't much help. Both cards have the
power connector plugged in, and both randomly give me static instead of a
dialtone, or an outside line.
Cards are 1 4 Port TDM400 FXO, and 1 4 Port TDM400 FXS. Computer is a
Compaq Proliant ML330
The full error is as
Since the irc channel wasn't any help, I will try posting my problem here.
I have two TDM400Ps less than a week old in a PC. All of the FXS ports
work fine, and all of the FXO ports worked fine up until thisafternoon. If
I try to dial in, I get a busy signal, if I try to dial out, all I hear
is a
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