Hello
I had an Asterisk installation working fine for CallerID on BT analog lines
using a Digium analog 4 port card. However, user switched to TalkTalk
without telling me and CallerID no longer works. However, if you connect a
UK CallerID capable phone into one of these analog lines directly
, 2006 12:37 PM
Subject: Re: [asterisk-users] CallerID in UK on TalkTalk - different to BT?
On Sat, Jul 08, 2006 at 10:59:49AM +0100, Angus Comber wrote:
I had an Asterisk installation working fine for CallerID on BT analog
lines
using a Digium analog 4 port card. However, user switched
Hello
I have been asked by a client to process a list of telephone numbers.
Asterisk should call each number in turn and if the recipient of the call
answers, play a message - eg from a wav.
How would I go about doing that?
Angus
___
--Bandwidth
Hello
I want to test asterisk with an H323 client. In Windows XP there is phone
dialer which can use H323. In Phone dialer I set H323 Line for phone calls
and Internet calls.
In Phone and Modem properties H323 provider I set:
H.323 gatekeeper: 192.168.0.20 (asterisk on my LAN)
Log on
Hello
I have setup a couple of sip accounts - here is my sip.conf:
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[200]
username=200
type=friend
secret=1234
port=5060
nat=never
[EMAIL PROTECTED]
dtmfmode=rfc2833
context=default
callerid=Angus 200
host=dynamic
insecure=very
group=1
than *8), because I read somewhere that for some reason Asterisk has a
problem with this feature and *8. It worked for us.
Alberto
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
Comber
Sent: Sunday, October 09, 2005 2:48 PM
To: [EMAIL PROTECTED
Hello
I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom
IP300 phones.
My config files look like this:
features.conf
pickupextn = *8
zapata.conf
context=frompstnisdn
group=1
callgroup=1
pickupgroup=1
I also edited sip.conf like this:
group=1
callgroup=1
pickupgroup=1
Yes sadly a typo on my part. It is pickupexten in features.conf
Any other ideas?
Angus
- Original Message -
From: Guido Hecken [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 12:54 PM
-f7db'
Any ideas?
Angus
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 1:31 PM
Subject: Re: [Asterisk-Users] *8 and group pickup not working
On Sun, Oct 09, 2005 at 12:32:12PM +0100, Angus Comber wrote
]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, October 09, 2005 2:35 PM
Subject: Re: [Asterisk-Users] *8 and group pickup not working
On Sun, 9 Oct 2005 21:32, Angus Comber wrote:
Hi
I have Polycom 600s and 500s but I find that we need
wireless generally struggles with brick walls.
- Original Message -
From: Matt Love [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, October 07, 2005 9:55 AM
Subject: RE: [Asterisk-Users] WiFi Phones
Hi
I
Your link doesn't seem to work.
Angus
- Original Message -
From: Cameron Steadman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 06, 2005 4:17 PM
Subject: [Asterisk-Users] Asterisk/Debian/VIA EPIA M Howto
I have written a step-by-step setup for
Hello
I am starting to learn AGI. I have setup an extension to play the
agi-test.agi perl script and the output I get is this on console:
On Polycom 300:
-- Executing Answer(SIP/200-72d2, ) in new stack
-- Executing AGI(SIP/200-72d2, agi-test.agi) in new stack
-- Launched AGI Script
Could you not just ignore the first answer and
watch out for the answer when the remote end picks up?
Angus
- Original Message -
From:
Chee
Foong
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, October 05, 2005 11:35
AM
Subject:
Hello
I want to setup a system where people can dial a number and then a system
will ask them questions for which they will leave answers. Eg something
like this:
Answer
Playback(whatisyournamemsg)
Record(yourname:gsm)
Playback(whatisyourheight)
Record(yourheight:gsm)
Playback(thankyou)
Hangup
, October 01, 2005 2:33 PM
Subject: Re: [Asterisk-Users] strange wave like noise on sip handset
On 9/30/05, Angus Comber [EMAIL PROTECTED] wrote:
On a Sipura SPA-841 handset (and also at other end) you hear a sea wave
like
sound - it gets louder then softer and continually repeats.
I don't
Hello
On business phones it is often possible to have call waiting (think that is
the feature) whereby if you are talking to a caller you can see another
caller has called and you can swap between callers. For example, to say
hello, I am on call with someone else now can I call you back.
Hello
I want to setup a system where people can dial a number and then a system
will ask them questions for which they will leave answers. Eg something
like this:
Answer
Playback(whatisyournamemsg)
Record(yourname:gsm)
Playback(whatisyourheight)
Record(yourheight:gsm)
Playback(thankyou)
Hello
In my extensions.conf file:
[frompstnisdn]
exten = s,1,Dial(SIP/200SIP/202,20)
exten = s,2,Voicemail(su200)
exten = s,3,Hangup
I use the s, start, extension to handle incoming calls.
In my zapata.conf:
context=frompstnisdn
This works ok on another asterisk box I setup. But on
I think the Asterisk must answer the call to be
able to then dial out on the second port. This is what happens on any
other PBX I have worked with in this sort of scenario. Is this a problem
for you?
Angus
- Original Message -
From:
Chee
Foong
To:
Hello
I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely
to be enough power for a 8 extension system with 6 external pstn lines?
How important is cpu? Is there some measure, eg xMHz CPU per extension or
something benchmark?
I have installed 512MB memory - again
Hello
We have setup a doorbell which has an inbuilt analog phone which is
connected to our Asterisk via a SPA2000 ATA. The problem we are getting is
that when a caller presses the buzzer it is taking two or more minutes to
finally call the reception phone.
In the SPA2000 I have set
asterisk-users@lists.digium.com
Sent: Friday, September 30, 2005 1:51 PM
Subject: Re: [Asterisk-Users] analog phone/door buzzer going through a
SipuraSPA2000 ATA dials really slowly
Angus Comber wrote:
Hello
We have setup a doorbell which has an inbuilt analog phone which is
connected to our
Hello
I am using a Snom 190 and the quality seems OK. Trouble is the volume is
quite low and full volume on the Snom is still too low. Is there something
I can do on the asterisk to increase the volume?
Angus
___
--Bandwidth and Colocation
Hello
On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like
sound - it gets louder then softer and continually repeats.
I don't remember hearing this when using other handsets. But what is this
effect? How can I reduce it?
Angus
-Users] Why does the s extension not work inmy
extensions.conf file
Angus Comber wrote:
Hello
In my extensions.conf file:
[frompstnisdn]
exten = s,1,Dial(SIP/200SIP/202,20)
exten = s,2,Voicemail(su200)
exten = s,3,Hangup
If you really want to use s, you will need to add an extension:
exten
Hello
When I dial out from my Asterisk (using Digium analog TDM04B card over pstn
line), calls appear to be from +34rest of number
I am in UK which is +44 so cannot work out why seeing +34.
In my zapata.conf I have:
loadzone = uk
defaultzone = uk
I can't find any country specific stuff in
? I assume you've tested the line with a normal
phone to make sure it's not a telco fault?
Ian
From: Angus Comber [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users
with a normal phone
to make sure it's not a telco fault?
Ian
From: Angus Comber [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cannot figure out why calls from my
could result in cracking the CPU. You should
never run a CPU without it's fan if it's meant to run with a fan. Even
if running it just as a file server. The fact that you are lucky
doesn't mean that you don't need a fan.
On 9/5/05, Angus Comber [EMAIL PROTECTED] wrote:
Hello
I am running Asterisk
Hello
I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series
motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink.
Currently system is running off standard IDE hard drive - because I couldn't
get astlinux to run with my Digium TDM04B card (only PCI
Does VoicemailMan have to be installed ? Why not available. I have setup a
mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup
up using *97.
My *97 code in extensions.conf:
exten = *97,1,Answer
exten = *97,2,VoicemailMain([EMAIL PROTECTED])
exten = *97,3,Hangup
PROTECTED] Em nome de Angus Comber
Enviada em: quinta-feira, 18 de agosto de 2005 16:58
Para: asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] asterisk seems to load but cannot connect
using -r?
I installed asterisk on SUSE 9.3. Stupidly I loaded selected to load
asterisk from the SUSE
?
On Fri, 2005-08-19 at 08:08 +0100, Angus Comber wrote:
Still get same:
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
The error message says it all. It thinks it's not running.
Check with the ps command.
--
Dave Cotton [EMAIL PROTECTED
It was my own stupid fault for installing the asterisk version available in
the SUSE distribution and then downloading and installing the latest
version. Another thing not to do!
Uninstalled old and re-installed asterisk and it worked!
Angus
- Original Message -
From: Angus Comber
When I attempt to compile the zaptel driver (latest CVS HEAD) I get this
compile error:
You do not appear to have the sources for the 2.6.11.4-20a-default kernel
installed.
make: *** [linux26] Error 1
When I load YaST I see kernel-source 2.6.11.4 as installed version. So why
do I get this
I installed asterisk on SUSE 9.3. Stupidly I loaded selected to load
asterisk from the SUSE DVD - then installed latest asterisk head using cvs.
At end of asterisk compilation mentioned modules in /modules where from
another installation.
My telephony cards working ok and if run asterisk
You could just add the line asterisk to /etc/init.d/boot.local
Angus
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 17, 2005 11:27 AM
Subject: [Asterisk-Users] Automatic
I have one Asterisk system working with a Junghanns BRI card and another
working with a Digium TDM card with an Intel D865 motherboard.
Angus
- Original Message -
From: jonny hashem [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 17, 2005 6:14 PM
Hello
I have installed a TDM04B and disabled any devices not required in my PC.
(TDM04B is analog card with 4 ports to plug into telephone co lines). I am
running this version of *
Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a i686
running Linux
As you see
Hello
I am (attempting) to run the astlinux version of Asterisk on a VIA embedded
platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup
OK. They worked fine with same card in traditional PC anyway.
I think need the module wcfxs for a Digium TDM04B card. Is this
it is some
config issue.
Angus
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 11:45 AM
Subject: Re: [Asterisk-Users] Module wcfxs - is it not part of astlinux?
On Sun, Aug 14, 2005 at 11:37:00AM +0100, Angus
it out.
Regards
Gurminder
On 8/7/05, Angus Comber [EMAIL PROTECTED] wrote:
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid=some name 300
auth=md5
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid=some name 300
auth=md5
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 05, 2005 7:39 PM
Subject: Re: [Asterisk-Users] Is this echo problem down to IP Phone
hardware?
Kris
Hello
I have a Grandstream GXP2000 with latest
firmware. When I use it holding the handpiece I don't hear any echo -
neither does other end. However, if I use it handsfree, the other end
notices echo when they speak - ie their voice is echoy. I hear their voice
being a bit echoy.
Is this
Hello
I have an application for Asterisk which could
involve potentially 5000 or more extensions. Possibly this number of
people making calls. All calls would be internal. Could enough
hardware be thrown at the problem to make this work? Anyone setup an
installation of this size? Any
Hello
A lot of my customers have people who are in the
office most of the time but occasionally wish to work from home. So they
may have a sip phone which is extension 208 in the office. When they work
from home they can of course plug in a sip phone into their broadband connection
and
I just wondered - might save me some development
effort!
Angus
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hello
I want to setup an Asterisk with three analog
lines. Two of the analog lines are the main office number. The other
line is the fax number. The fax machine plugs into the line 3 but also
will be a connection to the third port on a Digium analog card.
Reason for the third line into
Apologies for being a bit of a Linux
newbie...
I have got a working * system but each time I
reboot my box I need to:
modprobe qozap
ztcfg
asterisk
Now I realise this is really a Linux question but I
am struggling with the problem and any help would be much
appreciated.
There is a
Hello
I am using a Junghans quadBRI ISDN card and it is
loaded and working. In Asterisk if I connect to ISDN line it is detected
and tells me so.
In my zapata.conf I have
(abbreviated):
[channels]
switchtype=euroisdn
signalling = bri_cpe
context=default
group=1
channel = 1-2
;plus group
really do this.
JASON WALKER
- Original Message -
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11
AM
Subject: [Asterisk-Users] Should this
work?
Hello
I am using a Junghans quadBRI
From:
Angus
Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 25, 2005 8:11
AM
Subject: [Asterisk-Users] Should this
work?
Hello
I am using a Junghans quadBRI ISDN card and it
is loaded and working. In Asterisk if I conne
Hello
I am using a Junghanns QuadBRI ISDN card - the
module name is qozap. If I like at my interrupt assignment, qozap is
sharing interrupt 10 with libata and uhci_hcd.
I think libata is the IDE hard drive module and
uhci_hcd is a USB module.
linux:~ # modprobe qozaplinux:~ # cat
255.255.255.255
5060 Unmonitored
200 is a Grandstream GXP200 IP Phone and 202 is a
Grandstream BT100 IP phone.
relevant bit of sip.conf:
[200]username=200type=friendsecret=1234port=5060nat=neverdtmfmode=rfc2833context=defaultcallerid="Angus
Comber"
200host=dynamicdisallow=allallow=ulawallow
and returns 404
not found. That will explain why you can't call extension 777 from
extension 200. If you want to call extension 202, you will need to dial
202 on extension 200, not 777.
Regards,
Derek
- Original Message -
*From:* Angus Comber mailto:[EMAIL PROTECTED
a number via extensions.conf to
this address.
Have a look at www.voip-info.org and of course google.com to get to know
extensions.conf.
Regards,
Marc
Angus Comber wrote:
I think the 777 may be a bit of a Red Herring. I dialed 777 as a test.
I can't dial 202 from 200 if I actually dial 202
file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.
Did you create a dialplan for your specific configuration or did you just
copy the sample file?
- Original Message -
From: Angus Comber [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
extensions.conf
[default]
exten = _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
exten = _2XX,2,Hangup
just those 3 lines
do an 'extensions reload' in the CLI or just restart Asterisk
and see if it works
regards,
Mark.
On 7/25/05, Angus Comber [EMAIL PROTECTED] wrote:
I think the 777 may be a bit of a Red Herring. I
That was another problem - now fixed.
Thanks for all your help on extensions.conf
Angus
- Original Message -
From: Angus Comber [EMAIL PROTECTED]
To: Mark Edwards [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Sunday
Hello
I am sure this is a very basic Linux
question.
But every time I reboot my * I need to
modprobe module
and then
ztcfg
After doing this I can then run * without it
complaining about not loading a channel. The module being loaded is qozap
- a ISDN card.
What do I need to do to
, 2005 at 09:45:02PM +0100, Angus Comber wrote:
I am now getting this make error:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
Remove the line that includes asterisk.h
Hello
I have downloaded asterisk-addons but when I make
install get:
cc -fPIC -I../asterisk -D_GNU_SOURCE
-DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.capp_addon_sql_mysql.c:164:64:
macro "AST_LIST_REMOVE" requires 4 arguments, but only 3
On Thu, 2005-07-21 at 12:19 +0100, Angus Comber wrote:
Hello
I have downloaded asterisk-addons but when I make install get:
cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID
-I/usr/include/mysql -c -o app_addon_sql_mysql.o
app_addon_sql_mysql.c
app_addon_sql_mysql.c:164:64: macro
.
Thx MAG
Angus Comber wrote:
Hello I have downloaded asterisk-addons but when I make install
get: cc -fPIC -I../asterisk
-D_GNU_SOURCE -DMYSQL_LOGUNIQUEID
-I/usr/include/mysql -c -o app_addon_sql_mysql.o
app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro
+0200, Christoph Eicke wrote:
On Thursday 21 July 2005 15:28, Angus Comber wrote:
My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
It is a version put together by Junghanns.net - for working with their
ISDN
cards. Mmm I wonder if that is the problem? If so then what version
recording files for say a 10 user system? What about voicemail - I
suppose files could be emailed and deleted immediately?
Angus Comber
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
Personally I wouldn't bother with the Mahler
book. I bought it in the hope that it might be the panacea I was looking
for. It wasn't. If you read it you will recognise a lot of the
standard text you will see on Digium or other web sites.
If I had time I would write the book
myself.
I did
_DIGITAL'make: *** [channel.o] Error
1ASTERISK
installed.
Installation
finished.Is the
problem here the line: channel.c:41:31: asterisk/transcap.h: No such f
doesn't.
See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Angus Comber
|Sent: Monday, July 04, 2005 4:34 AM
|To: Asterisk Users Mailing List - Non
Hello
I am at extension 200 and I know there is a
voicemail message waiting. I dial *97 and am prompted for the
password. I enter 1234 which I have set as my voicemail password.
What can I do to troubleshoot?
Angus ComberItel Office Software Ltd5
Enmore GardensLondon, SW14 8RFTel: 020
?
Angus
- Original Message -
From: Angus Comber
To: asterisk-users@lists.digium.com
Sent: Monday, July 04, 2005 12:20 PM
Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my
passwordincorrect
Hello
I am at extension 200 and I know there is a voicemail message waiting
If I try dmesg - no mention of a Wildcard TDM400.
Sorry I am fairly new to Linux. In Windows I suppose I would run some
hardware program which came with the card to see if I could manually set
IRQ's etc. What should I be looking at now?
Please feel free to point me to a good book or
Hello
Here is what I find.
Any help would be greatly appreciated.
Angus
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, June 21, 2005 2:09 PM
Subject: Re:
.
Overall, if it works, lucky you, if not, Too bad.
Hard to support Digium and suggest others purchase such a product.
Best you look for other interfaces to Asterisk.
John Novack
Angus Comber wrote:
If I try dmesg - no mention of a Wildcard TDM400.
Sorry I am fairly new to Linux
Hello
Maybe a silly question, but after some searching
couldn't find answer. Is there a number I can dial to pickup and listen to
my voicemail messages on my SIP phone? I am used to eg dialling *17 to
pickup my voicemail messages on Avaya system?
Angus
zapata_additional.conf
;Include genzaptelconf configs#include
zapata-auto.conf
What am I doing wrong?
Angus Comber
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk
Hello
I am struggling to get my TDM04B working.
Just to rule out a hardware problem how can I check that the hardware
works? How can I then check that the drivers are loaded
correctly?
Angus
___
Asterisk-Users mailing list
section on the TDM400 on page 127 he says "In the UK, you
may need an adapter that provides a ring capacitor, or the phone may not
ring."
Can anyone confirm this. Also what is one of
those and where would I find a good supplier? I am in the trade so
wholesale would be OK.
An
no longer start
On Sat, Jun 04, 2005 at 11:20:47PM +0100, Angus Comber wrote:
This is what I have:
# Span 1: WCTDM/0 Wildcard TDM400P REV H Board 1
fxoks=1
fxoks=2
fxoks=3
# channel 4, WCTDM, inactive.
# Span 2: WCFXO/0 Wildcard X101P Board 1
fxsks=5
# Global data
loadzone = us
defaultzone = us
Hello
I want to setup an Asterisk in several offices with
4 BRI ISDN. I am looking for recommendations on hardware. Criteria
would be ease of setup, reliability and cost.
The Eicon 4 BRI cards seem fairly pricey.
Shame Digium don't do a ISDN BRI card.
Angus
Hello
I setup [EMAIL PROTECTED] with purely VoIP and it worked fine. I then added an
X100P card so I could call out / take inbound calls via PSTN and that went
fine. But I have just added a TDM400P card (specifically a TDM30B) and now
problems.
Here is some of the output. Any ideas on
] X100P installed OK,after added TDM400P
Asterisk would no longer start
On Sat, Jun 04, 2005 at 07:57:36PM +0100, Angus Comber wrote:
But I have just added a TDM400P card (specifically a TDM30B) and now
problems.
Found a Wildcard TDM: Wildcard TDM400 R Rev H (4 module)
wcfxs
Running ztcfg
anyone can see who is at the door.
Anyone aware of any suitable products.
Angus Comber
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
How does a Windows workstation fax via Asterisk? Has someone written a
Asterisk fax print driver? Or some other way?
Angus Comber
[EMAIL PROTECTED]
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hello
I have setup [EMAIL PROTECTED] and can login
to the system via the asterisk box. But if I try same username and
password to login using the Asterisk Management Portal I try the same username
and password and cannot login. says authorization failure. I have
tried from a Windows 2000
My home office is away from my house - so if anyone
rings door I cannot hear it. How would I rig up a doorbell which would
ring an extension on my Asterisk box?
Angus Comber
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users
88 matches
Mail list logo
