I am having problems with my zaptel channels on my fresh install of Asterisk
1.4.2 on Fedora core 6.
I have a new Digium TDM400P with 2 FXO modules.
Both dmesg and ztcfg -vvv show the FXO modules loading correctly:
-
Zaptel Version: 1.4.1
Echo Canceller: MG2
Configuration
Does anyone in New Zealand have any ATA devices in
stock I.e. Sipura SPA-2000?
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Are these phones any closer to being avaliable?
- Original Message -
From: Adam Hart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, April 01, 2004 5:38 PM
Subject: Re: [Asterisk-Users] Virbiage Phones - Vapourware??
Aaron Martin wrote:
Has anyone heard any more info
Has anyone managed to get Early-Dial working with
the grandstream phones?
On my older phones running firmware 1.0.3.X it
works fine, but it doesnt work on the newer versions..
Does anyone have any recomendations for a free
Windows softphone, SIP or IAX that supports the following features:
* Message Waiting Indicator
* Consultative Transfers
* Speed Dials
Is there anyway to display parked calls or call queues on this, so that
users can just drag and drop to connect to the calls?
- Original Message -
From: Nicolas Gudino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 8:52 AM
Subject: [Asterisk-Users] ANNOUNCE: Flash
Has anyone heard any more info about the Virbiage
FT201 VoIP phones?
About 3 months ago I was told they were 6 weeks
away, about 3 weeks ago I was told they were 2 weeks away, and now I am told
they are 2 months away again! Are they EVER going to arrive? Can
anyone shed some light on this?
Ok now that we have a Asterisk server running quite
well, we want to put it onto a more appropriate device, i.e. not a beige box
computer, but perhaps some kind of embedded linux appliance.
Has anyone already done this? Any suggestions
on some tidy, small, suitable linux systems to use for
I have Asterisk running with a combination of SIP
and H323 clients. I am using the OH323 module instead of the H323
one.
When the SIP clients ring each other, they can hear
a ringing noise in the ear peice to let them know that the other parties phone
is ringing. However, when the H323
Where / how do I set DTMF payload type to 101?
- Original Message -
From: Josh Roberson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 01, 2004 3:17 PM
Subject: RE: [Asterisk-Users] Re: Grandstream Early Dial
I've never had early dial working, however, I resolved my
caught up and that
QWest
screwed up their number porting. They moved their numbers from
QWest to
anohter provider and they aren't working... as of lastnight he was
about
to smack Qwest! :P
Just an FYI
bkw
On Wed, 3 Dec 2003, Aaron Martin wrote:
Sorry
I have upgraded my grandstream phone from firmware
1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the
older firmware earlydial worked fine with my asterisk server, but now as soon as
I have dialed the number I get a congested tone, and the number 4 flashes up on
the
: [Asterisk-Users] John Brown from Chagres!
Me too.
MATT---
PS- I apologize to all of those on the list that are sick of the Chagres
messages.
-Original Message-
From: Aaron Martin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 17, 2003 6:56 PM
To: [EMAIL PROTECTED]
Subject: Re
[For Google to find for others:]
I have a Micronet SP5052 FXO gateway which I
currently use in Peer-to-peer SIP mode with my asterisk box.
I have been having an intermittant problem with the
Micronet cutting off the incoming audio stream (i.e. PSTN to VoIP) so that the
user on the PSTN side
(and paid for) over a month
ago are going to be shipped to us. If you get a hold of him let me know,
I'm
still out 20 phones.
MATT---
-Original Message-
From: Aaron Martin [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 4:26 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
for the Grandstream phones
we ordered, but so far I have not had a single response, nor have the phones
arrived!
Please contact me ASAP
Aaron Martin
Comtek Computing Solutions Ltd.
I am attempting to contact John Brown from Chagres
Technologies, I know he watches this list. Please contact me ASAP John, I
have been trying to get hold of you for the last few weeks regarding an order
but so far havent had any luck!
Regards,
Aaron Martin.
5 - ringer volume
7 - 'message' button should flash for message waiting rather than LCD
5 - LCD backlight can be set to always on
3 - wall mount hook for handset
8 - ability to lock the menu on the phone to stop users from 'playing'
7 - speakerphone is not loud enough, even when turned up full
10
well, and el-cheapo consultative
working too ...
Might be better ...
On Mon, 2003-10-20 at 13:57, Aaron Martin wrote:
Hi Marcel,
Good to hear that everything is working well for you.
Just one question, how do your users transfer calls to each other? I.e.
is
it announced or blind
Hi Marcel,
Good to hear that everything is working well for you.
Just one question, how do your users transfer calls to each other? I.e. is
it announced or blind?
Regards,
Aaron.
- Original Message -
From: Marcel Prisi [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October
Also, what do the different 'format' numbers mean? Is there a table
somewhere showing which format is which number?
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 15, 2003 7:37 AM
Subject: [Asterisk-Users] use of SIP SHOW
I have also found the build quality to be very poor, i.e. I bought 3 phones
a few weeks ago, and when they arrived one of them had orange glue spots on
it, another had a hook that was sticky due to glue on it, and the other
didnt have any screws holding the motherboard in place!
- Original
:40 PM
Subject: Re: [Asterisk-Users] use of SIP SHOW CHANNELS question
On Sunday 19 October 2003 18:01, Aaron Martin wrote:
Also, what do the different 'format' numbers mean? Is there a table
somewhere showing which format is which number?
*CLI show codec 4
4 (1 2) G.711 u
Does anyone know of any reliable supplier for
Grandstream phones?
I tried dealing with David Li from Grandstream, but
after emailing him an order in August, and asking how he wanted payment, I never
got a reply...
James Ho from DGTimes was happy to give me pricing,
but when I sent him an
Hi Mark,
A switchboard type php application would be great, one that could show all
current calls / extensions etc and their states?
Just my $0.02
- Original Message -
From: Mark Evans [EMAIL PROTECTED]
To: Asterisk [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 9:04 AM
Subject:
I am trying to setup X-Lite with asterisk using the settings I found in the
archives:
http://lists.digium.com/pipermail/asterisk-users/2003-June/013597.html
My sip.conf has the following in it:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ;
For the information of others (and Google) this is
the table structure I used to get cdr_mysql working:
CREATE TABLE cdr ( calldate varchar(255)
NOT NULL default '', clid varchar(255) NOT NULL default '',
src varchar(255) NOT NULL default '', dst varchar(255) NOT NULL
default '', dcontext
Cant have been that obvious...
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 22, 2003 4:32 PM
Subject: Re: [Asterisk-Users] MYSQL Table Structure
On Monday 21 July 2003 23:21, Aaron Martin wrote:
For the information of others
Would this work with SIP / H323 phones??
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 1:11 AM
Subject: Re: [Asterisk-Users] Self parked but avaliable
On Sun, 2003-07-20 at 06:43, [EMAIL PROTECTED] wrote:
Is there
How do I use parking? I thought all I had to
do was hook flash, but this immediately cuts the other end of the call
off..
How do I make Asterisk provide these services?? I cant seem to get anything
to work..
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 11, 2003 2:46 AM
Subject: Re: [Asterisk-Users] H450 problems
Aaron Martin wrote:
Hello people
Hello people,
I am using Asterisk with a handful of Micronet
SP5100 IP Phones and a Micronet SP5052 FXO Gateway.
So far I have incoming calls ringing all the phones
correctly, outgoing calls working, voicemail working and calls between phones
working. The only think I cant get working is
] H450 problems
Hello Aaron Martin,
Yesterday we discussed the possibility to buy some Micronet hardware. We
haven't used hardware from this vendor yet and I wanted to ask you a
question about the Micronet SP5052 FXO Gateway features.
Is there any way to setup SP5052 FXO to answer a call from
I am trying to create an extension that will ring 2
ip phones if the receptionist doesnt answer. I have used the following
extension:
exten = 1401,1,Dial(H323/10.0.3.14,10,r)
exten = 1401,2,Dial(H323/10.0.3.13H323/10.0.3.14,10,r) exten
= 1401,3,Hangup
It appears to work well, i.e. it
The hardware we are planning to use
is:
Micronet SP5050 FXO Gateway
http://www.micronet.com.tw/Products/VoIP/SP5050.asp
Micronet SP5100 IP Phone
http://www.micronet.com.tw/Products/VoIP/SP5100.asp
We are hoping to use this hardware along with
AsteriskPBX to replace our aging PBX system.
Anyone in New Zealand using AsteriskPBX? If
so, what hardware are you using to connection to Telecom's lines?
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