Is anyone out there using any of the newer linksys phones since Cisco
took over? I am more specifically looking at the spa-941 942's. Just
curious about call quality, programability, and functionality with asterisk.
I have read through the literature, but would like some real world feedback.
Dr. Michael J. Chudobiak wrote:
Hi all,
I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they
work, but sometimes the caller just gets dead air or disconnects. IAX2
debugs show HANGUP and INVALID codes in these cases, rather than a
proper RINGING transaction.
My firewall is
Sean Cook wrote:
OK... maybe I got a little anxious and ran out and bought a Tyan GX28
with dual Opteron (dual core) processors. (It is a nice server ;) ) I
did neglect to find out that you can not manually set the IRQ's on this
motherboard. I am now stuck sharing an IRQ with the ethernet
Is there a list or matrix somewhere that shows what codec can be
transcoded? I am playing with different allowed codecs between my
asterisk box and some of my providers testing voice quality and
bandwidth usage on my cable connection, and I occassionally run into an
issue where asterisk cannot
Hi all,
I just installed openSUSE 10.0 on a spare machine to
try and do some development work. I did a checkout on
libpri, zaptel, and asterisk and everything compiled and
installed perfectly. My issue is with the zaptel script
placed in the rc.d directory to automatically initializ
the
On Tue, 14 Mar 2006 14:32:02 +0100
Olle E Johansson [EMAIL PROTECTED] wrote:
14 mar 2006 kl. 13.35 skrev Matt:
Right saw that. But I'm trying to get away from using
CVS-HEAD :)
We all are. Every developer have switched from CVS to
Subversion :-)
This is not the development branch, but
On Tue, 14 Mar 2006 13:44:57 -0500
Matt [EMAIL PROTECTED] wrote:
http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/
Thank you I was looking directly under asterisk and
not team. :-)
Robert
___
--Bandwidth and Colocation
On Tue, 7 Mar 2006 09:12:25 -0700
Douglas Garstang [EMAIL PROTECTED] wrote:
I have a configuration where RTP traffic is going out
interface pub0, and coming back into through pub1.
I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an
shows:
udp0788 0.0.0.0:5060
Sorry, this is off topic to asterisk itself, but is about
the list server.
I had a power failure lastnight at home, where my email
server resides, and my network was down for about 20
minutes, that was after 45 minutes of uptime on UPS. Since
power was restored, around 9:45 PM EST on 2/16,
On Wed, 15 Feb 2006 08:59:22 -0800 (PST)
housi mueller [EMAIL PROTECTED] wrote:
Hi there,
I would like to connect an Aasterisk Server with a
Panasonic PBX (has E1extension).
I only need 4 Lines. So I thought I could use an
Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1
card
I sent the below message out last Friday when the list
seemed to be having issues. Never got any responses and
not sure if it just no one knows or if it did not get
through.
Please don't flog me too bad for reposting... :-)
Hi
Hi all,
I am having some audio quality issues with a provider
under sip. The issue I am having is that the audio seems
to be acting like a simplex connection. I have tested my
setup with a second provider and the audio quality to them
is great. Checked network type issues, latency, packet
On Wed, 11 Jan 2006 15:38:04 +0100
Matt Riddell (IT) [EMAIL PROTECTED] wrote:
I would like to develop a video file player tool inside
Asterisk. When
calling to an extension answer and Play a video file
(H264). With the
applications PlayBack is not possible to give a video
extension (only
On Wed, 11 Jan 2006 11:39:20 -0700
Douglas Garstang [EMAIL PROTECTED] wrote:
Peter,
Too slow! We're going to potentially be doing several
MySQL lookups for routing even the most basic of calls,
and if every one of those queries has to make a call out
to an AGI script, it would become a
On Tue, 13 Dec 2005 02:25:50 +0100
Patrick [EMAIL PROTECTED] wrote:
On Mon, 2005-12-12 at 22:08 -0300, Juanjo Portela wrote:
My cdr_mysql.conf is the same I was using for
version.1.0.9 and it is as follow
[global]
hostname=localhost
dbname=dbasterisk
password=dbpassword
user=dbuser
On Fri, 09 Dec 2005 00:36:18 -0500
Matthew matthew@zeut.net wrote:
Hello, has anyone taken their cell phone number and
ported it over to a voip provider? If so, what voip
provider and what was your experience?
Matt
Matt,
I have done this. I had a cell number with ATT
Wireless and
on the outside that is registering back into your Asterisk
box.
I know someone will correct me if I am wrong, but I
believe that is the way it works. You have to forward
ports for SIP because of the way the RTP stream is setup.
Robert Webb
___
--Bandwidth
Hi all..
I just setup a test box with Debian running kernel 2.6.
Went to CVS and did a checkout of the new beta 2 release
using the command: cvs checkout -r v1-2-0-beta2 zaptel
libpri asterisk asterisk-addons asterisk-sounds.
I then compiled libpri fine and moved on to zaptel. Did a
make
On Tue, 01 Nov 2005 10:18:45 -0500
Paul Zimm [EMAIL PROTECTED] wrote:
I then compiled libpri fine and moved on to
zaptel. Did a make clean then make install and get the
following error:How about `make linux26' ?
I am not up to speed on make or its
I have just a quick setup question about how some of you
have hardware setup.
Basically, for a system that has an average volumes of
calls in an office setting, are you using one or two
network cards. I am just wondering if it owuld be any
advantage to having one NIC for the extensions and
On Fri, 21 Oct 2005 10:25:59 -0400
Paul [EMAIL PROTECTED] wrote:
Kanuri, Seshu (Company IT) wrote:
[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.
Olle is right. Even if it is a free service it does not
belong here.
This forum is for any Asterisk related
Just tested mine and it is working
fine.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake
KroneSent: Thursday, October 20, 2005 5:32 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Goiax.com DID not working anymore?
I have a TDM card with one FXO and one FXS. I am trying to make sure I
understand correctly the TX and RX Gain in the Zapata.conf correctly. If
I have a phone cord plugged into an FXO port tied into a POTS line and
boost the TXGain, am I correct in thinking that the audio going back to
the phone
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gary Smith
Sent: Monday, September 05, 2005 5:13 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM11B pinout
Hi Asterisk Community,
I have a development with a
On Wed, 24 Aug 2005 14:47:25 -0400
Araba, Michael [EMAIL PROTECTED] wrote:
Thanks John, You are my savior. This is such a great
relief. Apparently
realtime will not use either '127.0.0.1' or 'localhost'
to connect to the
database. I had to use the actual IP address attached to
the NIC before
On Wed, 24 Aug 2005 15:25:15 -0400
John Novack [EMAIL PROTECTED] wrote:
In my case, mysql is set to any host
So, yes, it does seem to be an Asterisk issue
And my buddy is pretty savvy with mysql, Linux and
databases on Unix/Linux, having worked for a large IT
company for some 20 years.
SNIP
Ok, I figured it out, * was not using the config under
the [router]
context in the config file. Once I enabled g729 in
[general] it worked.
So the question is why does * ignore this config for the
192.168.77.254
endpoint?
in sip.conf:
[router]
type=friend
context=default
On Wed, 27 Jul 2005 18:07:23 +0200
Walid Azab [EMAIL PROTECTED] wrote:
Hi..
I am trying to do something but it is giving me some
hard time here. I have
an IAX2 trunk to FWD which is registered and working
just fine. I have =
011|. as my dial pattern to allow that. But if I want to
dial a
On Tue, 26 Jul 2005 10:24:20 -0400
Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Tuesday 26 July 2005 09:43, chouck wrote:
I assure you I have read the asterisk handbook many
times. The
immediate=yes is for picking up a phone on an fxs and
having it immediately
dial an extension. I am
On Mon, 25 Jul 2005 15:44:07 +0200
Alexis F. [EMAIL PROTECTED] wrote:
Hi,
I would like to use a digum card to call an external
number through my PSTN. I think that I have a problem in
the configuration. Asterisk returns me app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
On Wed, 20 Jul 2005 18:00:24 +0200
Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I spot weird behaviour of latest Firefly 3rd party on my
laptop. Sometimes it comes to state that it won't start
(hangs on Initializing ) and it again works after
system restart... Didn't yet figured out how to
SNIP
.
And please note that in general members of the list
dislike List Police even more than they do off-topic
posters.
B.
Cool...
I will be sure to ask any question I have now and
expect not to get Policed by anyone on this list. Sounds
like this is the list for the support of
On Thu, 7 Jul 2005 10:49:32 -0700
Dan Adams [EMAIL PROTECTED] wrote:
Hi, I am sorta a newbie to the asterisk community at
least in the realm of
hardware types. I was wondering, what type of card is
used to allow asterisk,
on a slackware installation to talk to a standard phone
line so that
On Mon, 4 Jul 2005 20:56:30 -0400
Jimmy Smith [EMAIL PROTECTED] wrote:
6 beyond-the-network.LosAngeles.savvis.net
(208.173.57.30) 33.966 ms
34.143 ms 33.841 ms
7 * * *
hangs there...
savvis invoice paid ?
beyond-the-network a black hole ?
On 7/4/05, Gary Reuter [EMAIL PROTECTED]
On Tue, 5 Jul 2005 22:07:11 +0800
Ian Bert Tusil [EMAIL PROTECTED] wrote:
I've just Installed [EMAIL PROTECTED] i browsed it's
built-in AMP. it
prompts for a login if you click on asterisk management
portal. i
tried
user:[EMAIL PROTECTED]
pass:password
and
user:admin
pass:password
but
On Tue, 05 Jul 2005 11:26:39 -0700
Bruce Ferrell [EMAIL PROTECTED] wrote:
I've gotten word from their Marketing VP. They are
doing some kind of massive move and expect to be down
until Thursday
Sounds like their Marketing VP needs to get a clue and let
customers know what is going on.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Keith Caldwell
Sent: Saturday, July 02, 2005 8:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Radon
Sent: Saturday, July 02, 2005 10:49 AM
To: andrew matthews; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] passing through MWI info
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tom Rymes
Sent: Friday, July 01, 2005 11:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP/[EMAIL PROTECTED] (asterisk at home) custom
On Fri, 01 Jul 2005 11:10:27 -0700
Chris A. Icide [EMAIL PROTECTED] wrote:
John Novack wrote:
Mike Myers wrote:
snip
Wow, this is a serious problem for me. I don't need
to actually check the voicemail itself from Asterisk,
just to be able to tell that there is voicemail
waiting. Are
See messages inline...
On Thu, 30 Jun 2005 09:48:51 -0700 (PDT)
David [EMAIL PROTECTED] wrote:
Hi,
I am trying to do the world's most simple install.
I have a Wildcard TDM400P with 3 ports: 1 FXS on port
1 and 2 FXOs on ports 3 and 4. (i'm not using port 3
for now, put want it for expansion
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of hank
Sent: Thursday, June 30, 2005 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Gizmo: Skype done right?
they claim to have a
On Wed, 29 Jun 2005 08:15:20 -0400
Chris Mason (Lists) [EMAIL PROTECTED] wrote:
An ethereal trace indicates the IP address is active, but
it is not
responding to iax packets (registration). So, either
their asterisk
app has failed or they have folded their tent as well.
I am sure it's
On Wed, 29 Jun 2005 13:56:00 -0700 (PDT)
chawki hammoud [EMAIL PROTECTED] wrote:
Is there a way to hide the callerid on analog line on
outgoing calls. Any ideas whether it could be done
through configuration, a script or hardware.
Thanks;
It would have to be done through who ever provides
On Mon, 27 Jun 2005 15:27:22 -0400
Andrew Kohlsmith [EMAIL PROTECTED]
wrote:
On Monday 27 June 2005 14:31, Michael Di Martino wrote:
If this list spent at least half the time on helping
other asterisk
admins as it does on
trivial things like LiveVoips bankruptcy it just might
be a great
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Di Martino
Sent: Monday, June 27, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt - Why this thread
I agree with
I am trying to get an iaxy device to connect to my asterisk box over the
public cloud however
It fails register and I cannot figure out why. Below is my iax.conf,
iaxy setup file and out from iax2 debug.
My iax.conf
[u7403]
type=friend
accountcode=iaxy
host=dynamic
secret=u7403p
context=from-iaxy
On Fri, 24 Jun 2005 09:27:45 -0400
Michael Di Martino [EMAIL PROTECTED] wrote:
Ok I have added the timeout value but it still does not
pick. However
jus to test voicemail function
I comment out the first line and voice does pick up.
What could be
wrong.
exten = 7403,1,Dial(IAX2/u7403/1/5)
On Fri, 24 Jun 2005 13:10:13 -0400 (EDT)
[EMAIL PROTECTED] wrote:
On Friday 24 June 2005 11:02, [EMAIL PROTECTED] wrote:
We are on a real world... Every cyber cafe has its own
little
hacker/cracker that is sniffing out... A simple ethereal
capture could
give me a bank pin number... It is
On Wed, 15 Jun 2005 14:37:42 -0400
Huddleston, Robert [EMAIL PROTECTED] wrote:
Site down again?? Voip-info.org? or maybe really slow?
Up here for me at 15:00 EDT...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Kohlsmith
Sent: Saturday, June 11, 2005 11:58 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] ATTN: Keith
On Saturday 11 June 2005 11:35, Tracy Phillips wrote:
That
running the iax ping
tool found here:
http://www.voip-info.org/tiki-index.php?page=IAX
It is called IAX Ping tool 1.01 for windows. That will
confirm that the two ends can talk IAX.
Robert Webb
___
Asterisk-Users mailing list
Asterisk-Users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Wolfe
Sent: Friday, May 27, 2005 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue
Maybe I should my pictures in with me
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Quintin
Sent: Thursday, May 26, 2005 1:15
PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users]
[EMAIL PROTECTED] - mysql login
Hi all,whats the root password for [EMAIL
And to point out additional info, the backlight for the entire phone
flashes when the mailbox it is programmed to monitor has a message. MUCH
easier to see than a little flashing red light.
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wilson Pickett
Sent: Wednesday, May 25, 2005 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Digium FXS modules too fragile?
SOME people also
It is called search the wiki!!!
http://www.voip-info.org/wiki-Asterisk+Zap+channels
But can only be done for ZAP channels..
On a side note.
When are you guys going to fix your QWEST peering out of Richmond??? I
would really like to be able to use my Asterisk box during business
hours. But a
I was just browsing Digium's web site and noticed they are
taking orders for the new IAXy. Has anyone purchased and
tested one of these yet?? I have thought about buying one
for testing, but want to make sure it isn't going to be a
flop like the last one.
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mike Clark
Sent: Wednesday, May 18, 2005 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Outbound dialing issue with FXO
We are installing a number of
On Fri, 13 May 2005 07:59:09 -0700
Wiley Siler [EMAIL PROTECTED] wrote:
Almost positive iLBC is not allowed Use uLaw...
They do allow for iLBC. From their FAQ page:
Codecs. Carriers with primarily business customers should
use the G.711 codec when sending VoIP traffic to VoipJet.
This
Hi, Is there a script in amp for adding the extensions? And can it be
modified? When adding a new extension it rewrites all of the
information it the context blowing out my additions.
You my want to try the AMP forum. Since they are the producers of AMP,
they may have a little better info.
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
Guys
I have a problem getting a TDM400P card to go.
It has 4 FXS ports (green modules) and I get this error:
[EMAIL PROTECTED] root]# ztcfg -v
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02:
FXS Kewlstart
Zttool shows nothing inside thebox.
I tried removing the x100 cards, moving the tdm card around,
disabled all usb and unnecessary stuff still, kudzu when
booting up shows the card and the card shows up on
/etc/sysconfig/hwconf but I wonder why it shows 2 of these
and I only have 1 tdm400p
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Friday, April 29, 2005 1:50 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Problems with TDM400P card
How do I remove it from kudzu?
I am
Ok,
So I am trying to still figure out my ringing issues. This time I
grabbed the butt set I own and hooked it into my pots line. With the
butt set in monitor mode, I called the pots line so I could actual hear
the AC ring. It was a low frequency ringing sound like I am accustomed
to.
I then
On Thu, 28 Apr 2005 16:01:44 +
Luz Lopez [EMAIL PROTECTED] wrote:
Hi All.
I have installed Asterisk on linux Redhat version 9, I
follow step by ssstep the installation, my card digium is
TDM400P, whith modprobe wcfxs I have load this module.
My vonfiguration files are in /etc/asterisk, the
On Wed, 27 Apr 2005 11:24:24 -0600
Andrew Elchuk [EMAIL PROTECTED] wrote:
Hi,
I have two of the above installed into a server running
Asterisk on Debian Linux. Currently, only two phone
lines are connected to the system. I had both phone
lines plugged into the one card, and it worked fine
On Wed, 27 Apr 2005 13:02:56 -0500 (CDT)
Paul Shiflet [EMAIL PROTECTED] wrote:
I just received my TDM400 card from digium with 2 fxo
and 2 fxs
interfaces. They are all RJ45 ports as opposed to RJ11
like my POTS
phones. How do i interface my POTS phones with this; can
i just crimp an
RJ45
SNIP
#user_info: phone
# SIP Configuration File (stop)
When the phone tries to register, all I get in the Asterisk
console is this:
Apr 23 08:22:29 NOTICE[26568]: chan_sip.c:8804
handle_request_register:
Registration from
'sip:[EMAIL PROTECTED];user=phone'
failed for '24.18.147.95'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Webb
Sent: Saturday, April 23, 2005 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
List Receiver
Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device
On Fri, 22 Apr 2005 10:37:32 -0400
Mark Phillips [EMAIL PROTECTED] wrote:
I have a full PRI installed on my * machine. I can get
inbound calls just fine but can't make outbound ones.
Zaptel.conf says;
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf says
language=en
context=default
On Fri, 22 Apr 2005 11:48:18 -0400
Mark Phillips [EMAIL PROTECTED] wrote:
Nothing happens. I get the same (non)error.
I get plenty of output when receiving a call however.
Mark
Andrew Kohlsmith wrote:
On April 22, 2005 10:41 am, Robert Webb wrote:
Your zapata.conf should look like this:
language
SNIP
== DISCONNECT_IND PLCI=0x101 REASON=0x3481
== No one is available to answer at this time
How changing from CAPI to a zaphfc card will correct
this error I don't
know, and problably neither does the person who
suggested it.
REASON 0x3481 is Unallocated (unassigned) number. =
Wrong
On Fri, 22 Apr 2005 01:26:45 +0800
Nathaniel Angelo A. Torres (247talk)
[EMAIL PROTECTED] wrote:
Hi, here's the content of my Zapata.conf
[channels]
language=en
context=default
signalling=em_w
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
On Wed, 20 Apr 2005 10:24:37 -0500
Josiah Bryan [EMAIL PROTECTED] wrote:
On Wednesday 20 April 2005 10:29 am, David Choo wrote:
Dear All,
My boss has placed a requirement for me to forward all
our IDD calls
through a partner's IDD service, which requires us to
call a 8 digit
number, wait for 1
On Wed, 20 Apr 2005 18:33:44 +
Jaime Blanco [EMAIL PROTECTED] wrote:
Hi,
I just installed the asterisk and the X100P card. I can
receive calls from PSTN and it can ring on a Grandstream
SIP Phone. From the SIP Phone I can dial the demo
extension on asterisk pbx. The issue is as soon as
See inline responses...
On Mon, 18 Apr 2005 10:43:30 -0400
Ian Pattison [EMAIL PROTECTED] wrote:
I don't know how everyone else is doing but my woes are
continuing.
Hardware:
Digium TDM400P (REV G according to the silk screening on
the board) 2xFX0, 2xFXS purchased in August/September
2004
On Mon, 18 Apr 2005 11:54:09 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
Inline...
Hi, I did not find any useful information to configure a
Wildcard
TDM400P with a FXO card. I've tried everithing, I tried
configure it
using the cvs and the information from digium page, I
tried to
configure it
Senerio
multiple * boxes connecting to a central * box with T1 card via IAX2.
1box 1 abd 2 work fine all the time
box 3 - after approx 10-15 minutes with no calls - central box with T1
card
fails to deliver incoming calls to box 3.
Connectivity is good, * exten-2-exten good
Ok, I am
On Thu, 14 Apr 2005 08:14:37 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
I was following a discussion on this list about the
TDM400P
revisions.
It is my understanding that the current revision that
one
should have
is the Rev. H and not the E/F. I have not yet been
able to
verify
On Thu, 14 Apr 2005 07:19:50 -0700
Sean Kennedy [EMAIL PROTECTED] wrote:
G.Marshall wrote:
Hello,
I can not find anything on this, so it may not be
possible.
I would like to dial one number which then rings at least
two extensions
at the same time. Not a hunt group, but ringing at the
On Thu, 14 Apr 2005 10:59:11 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
I was following a discussion on this list about
the
TDM400P
revisions.
It is my understanding that the current revision
that
one
should have
is the Rev. H and not the E/F. I have not yet been
able to
On Thu, 14 Apr 2005 12:45:44 -0400
Ian Pattison [EMAIL PROTECTED] wrote:
My specific issue has to do with ringing on my FXS
ports.
A Northen Telecom Harmony phone (circa 1983) rings
normally but when I connect my newer GE 2.4GHz cordless I
never get more than 1/2 ring (it lights up and works
On Thu, 14 Apr 2005 11:42:34 -0700
Sean Kennedy [EMAIL PROTECTED] wrote:
Hi all
With the recent thread on line presence in asterisk, can
anybody tell me if there is a phone out there that
supports this? Say I have 20 extensions: Is there any
way, hardware based, for me to see the activity on
On Mon, 11 Apr 2005 10:54:30 -0400
Robert Webb [EMAIL PROTECTED] wrote:
Good morning all..
I was following a discussion on this list about the TDM400P
revisions.
It is my understanding that the current revision that one
should have
is the Rev. H and not the E/F. I have not yet
On Tue, 12 Apr 2005 15:04:26 -0400
David Brodbeck [EMAIL PROTECTED] wrote:
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
I don't think the GPL obliges you to give credit to
anybody.
In fact, I think that's a key difference between the GPL
and the BSD
license.
On Tue, 12 Apr 2005 14:05:06 -0500
mr. barker [EMAIL PROTECTED] wrote:
I am using [EMAIL PROTECTED]
When I manually add anything to the
extensions_additional.conf file it gets
rewritten when I add an extension using the web
interface
I am trying to include the monitor function .. I got
that
Good morning all..
I was following a discussion on this list about the
TDM400P revisions. It is my understanding that the current
revision that one should have is the Rev. H and not the
E/F. I have not yet been able to verify the rev stamped on
the board, but zaptel is reporting that I have
Sorry for the initial no subject line. Was in a hurry when
I typed this and somehow missed putting it in.
Please accept my apologies
On Mon, 11 Apr 2005 10:54:30 -0400
Robert Webb [EMAIL PROTECTED] wrote:
Good morning all..
I was following a discussion on this list about the
TDM400P
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dean collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay listing selling
Asterisk @
I was following a discussion on this list about the TDM400P
revisions.
It is my understanding that the current revision that one
should have
is the Rev. H and not the E/F. I have not yet been able to
verify the
rev stamped on the board, but zaptel is reporting that I
have the Rev.
Robert Webb wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Monday, April 11, 2005 5:35 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: Ebay
SNIP
If you look at a 'iax2 debug' log you will see things like:
Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF
Subclass: 6
Timestamp: 15832ms SCall: 2 DCall: 00167
[217.160.244.186:4569]
which seem to indicate the codes are making to my local asterisk
box,
or
On Fri, 1 Apr 2005 16:42:54 -0500
Kellner, Peter [EMAIL PROTECTED] wrote:
I've got an asterisk server 1.07 with a Digium TMD400P
(2fxo;2fxs). I
have it configured to answer an incoming line and
transfer to one of the
2 fxs's and it works.
I have noticed that on incoming calls I get
On Thu, 31 Mar 2005 10:27:24 -0800
hank smith [EMAIL PROTECTED] wrote:
isn't [EMAIL PROTECTED] included in 1.07? of asterisk? also
I checked the asterisk.org site and saw 1.06 but not the
latest when was it put up on asterisk.org?
Huh??? Last time I checked, [EMAIL PROTECTED] was an install
On Wed, 30 Mar 2005 08:29:39 -0500
Matt [EMAIL PROTECTED] wrote:
Hi,
What happened to asterisk @ home 0.7 that the
dialout-default macro no
longer works?
___
EVERYONE
This is NOT the [EMAIL PROTECTED] list group.
Please go to:
On Tue, 29 Mar 2005 12:55:41 -0600
Jeffrey Sharpe [EMAIL PROTECTED] wrote:
Thank you!
Jeffrey
Please do a little searching of the list next time. I just
answered this same question about 4 days ago!!!
Robert
___
Asterisk-Users mailing list
On Tue, 29 Mar 2005 12:30:31 -0800
Noah Silverman [EMAIL PROTECTED] wrote:
hi,
We are using PTSN lines connected through the Digium FXO
modules for our
incomming lines
When a caller calls in, the prompts play back at a
really high volume.
They are a bit distored and fuzzy since they are so
On Mon, 28 Mar 2005 12:24:00 -0500
steve szmidt [EMAIL PROTECTED] wrote:
On Monday 28 March 2005 12:19, Jon Walsh wrote:
How does one downlaod the upgrade only is there the
ability to do so
from the software or do you need to re-burn an iso or is
the iso an
upgrade version or the whole install
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