Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
See inline below with NB...
On Sat, Jan 9, 2010 at 11:56
On Sun, Jan 10, 2010 at 12:25:10AM -0800, Nitin Bahadur wrote:
Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'_X.' = 1. Dial(Zap/1)
[pbx_config]
-= 1 extension (1 priority) in 1 context. =-
jserver*CLI dialplan show default
[
On Sun, 2010-01-10 at 00:25 -0800, Nitin Bahadur wrote:
Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
That would suggest the card is looping the line (busying it out).
On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote:
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'_X.' = 1. Dial(Zap/1)
[pbx_config]
-= 1
On Sun, Jan 10, 2010 at 1:48 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote:
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal'
Hi,
I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing
a fast clicking sound and now I am not hearing any dial-tone. The FXO
card has 2 slots: [phone | line ]. I hve connected the wall-phone-input
to the line slot and phone to my home-phone. I do not hear any dial-tone
on my
On Sat, Jan 09, 2010 at 11:33:06PM -0800, Nitin Bahadur wrote:
Hi,
I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing
a fast clicking sound and now I am not hearing any dial-tone. The FXO
card has 2 slots: [phone | line ]. I hve connected the wall-phone-input
to the
Thanks Marco, I have installed Elastix 1.5.2. Elastix detect and configure
OSLEC.
Regards,
GM
a.. Subject: Re: TDM2400P dial tone is not present on phones, but the phone
ring with incoming calls
b.. From: Marco Sambo derwid...@x
c.. Date: Thu, 16 Apr 2009 07:43:36 +0200
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a call
but only with pulse dialing (not tone dialing) without hear sounds from the
headset. When I receive a call, I can to establish a communication, but without
hear sounds from the headset. When I dial any phone
Hi, excuse me, but I see in your code that you configure DAHDI with OSLEC.
How do you do? Which version you have installed?
Thank you.
Marco
2009/4/16 Giovanni Magallanes gmagalla...@gmail.com
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a
call but only
. . .
What version of asterisk is it?
1.4.4
What is the output of:
cat /proc/zaptel/*
Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1
Span 2: WCTDM/0 Wildcard TDM400P REV I Board 1
1 WCTDM/0/0 FXOKS
2 WCTDM/0/1 FXOKS
3 WCTDM/0/2 FXSKS
4
On Saturday, January 24, 2009, j...@j4computers.com wrote:
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial
plan and all LEDS on the card lit, I get no dial tone, plugging an
analog phone into ports 1 or 2, only a buzz and
On 1/25/2009 at 6:38 AM, j...@j4computers.com j...@j4computers.com
wrote:
. . .
What version of asterisk is it?
1.4.4
What is the output of:
cat /proc/zaptel/*
Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1
Span 2: WCTDM/0 Wildcard TDM400P REV I Board 1
1
It is giving a dial tone now. I am not quite sure what fixed it.
I did remove a typo or two (gasp!) from the zapata.conf file and had to remove
the ; as there were some complaints about that.
A reboot and restart seemed to . . . oh, wait, I also swapped the board's power
connector for
On Saturday 24 January 2009, j...@j4computers.com wrote:
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan
and all LEDS on the card lit, I get no dial tone, plugging an analog phone
into ports 1 or 2, only a buzz and click.
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and
all LEDS on the card lit, I get no dial tone, plugging an analog phone into
ports 1 or 2, only a buzz and click.
zaptel.conf -
defaultzone=us
loadzone=us
fxoks=1,2
On Sat, Jan 24, 2009 at 06:38:58PM -0500, j...@j4computers.com wrote:
This is, hopefully, just a case of brain fade.
With zapata.conf and zaptel.conf in place, asterisk loaded, no dial
plan and all LEDS on the card lit, I get no dial tone, plugging an
analog phone into ports 1 or 2, only a
Mojo with Horan Company, LLC wrote:
Just to be clear, I thought that dialtone provision didn't require the
power cable, just generating ring voltages? Can anyone say?
The DC-DC converter on the FXS modules supplies both ringing voltage and
line voltage. If the power connector is not plugged
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made
modprobe wctdm the fxs modules is lightened but there is no dial tone came from
it .
Can i get some help please.
Best Regards;
Wassim
_
Windows Live
On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i
made modprobe wctdm the fxs modules is lightened but there is no dial tone
came from it . Can i get some help please.
do you have the power cable
On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when
i made modprobe wctdm the fxs modules is lightened but there is no
dial tone came from it .
Once you've loaded the wctdm kernel module, you should get battery on
the
On Wed, Sep 05, 2007 at 10:27:48AM -0400, Jared Smith wrote:
On Wed, 2007-09-05 at 14:09 +, wassim darwish wrote:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when
i made modprobe wctdm the fxs modules is lightened but there is no
dial tone came from it .
Once
Just to be clear, I thought that dialtone provision didn't require the
power cable, just generating ring voltages? Can anyone say?
Moj
Anthony Messina wrote:
On Wednesday 05 September 2007 09:09:25 am wassim darwish wrote:
Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my
Date: Wed, 5 Sep 2007 09:21:19 -0800
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] No Dial tone came from fxs modules Just to be clear, I
thought that dialtone provision didn't require the power cable, just
: dinsdag 19 juni 2007 17:03
To: Lee Jenkins
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play dial tone withou answer
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed on the outbound leg.
Dave
On Tue, 2007
] On Behalf Of David Boyd
Sent: dinsdag 19 juni 2007 17:03
To: Lee Jenkins
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play dial tone withou answer
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed
Hi,
I'm looking fore a way to play a dial tone before our IVR platform
answered the phone line.
I want to use for the following reason:
When a caller calls our Voice Platform, the call will direct dial out to
a number.
I want to dial out before the inbound call is answered.
But now
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel back to the
originator
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed on the outbound leg.
Dave
On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote:
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you
New system install.
At what point, in bootup, should I be able to get a dial tone on the phone
ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug
into g.
At boot up, as soon as wctdm is loaded, all the ports go green, yet I do not
get a dial tone on the phone
On Wed, Mar 14, 2007 at 07:53:58AM -0400, joe acquisto wrote:
New system install.
At what point, in bootup, should I be able to get a dial tone on the
phone ports on a tdm400p? There are two fxo and two fxs ports. I
know which to plug into .
Only after:
* The card's driver has loaded
Thanks. Very reassuring. It really must be too early. g
joe a.
Tzafrir Cohen[EMAIL PROTECTED] Wrote: 3/14/2007 8:23 AM:
On Wed, Mar 14, 2007 at 07:53:58AM -0400, joe acquisto wrote:
New system install.
At what point, in bootup, should I be able to get a dial tone on the
phone ports on
Hi, i'm having problems with DTMF, the problems are with established
connections and some IVRS.
When i call to other number which has an IVR, some digits doesn't
work. I digit a long number (required by the IVR, at least a 10 digit
number) and it doesn't work. I think it's about DTMF signalling,
Maybe of you guys know the answer to this:
We have T1's that come from both MCI and Global Crossing as channelized (24
Ports per T) with inband (DTMF) delivery
of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4,
AMI, SF and Wink Start and so is Asterisk. I've moved these
hello asterisk users,
i an using asterisk cvs 1.0.9 in a pIII 733mhz 256MB RAMredhat 9.
i have a TDM400P with 2FXO and 2FXS modules. in my fxs i want to get australian dial tone and for all asterisk operation i want to use australian tones. by default it is US. to change this i have edited
I'm seeing all sorts of problems and it's probably more of my lack
of experience than anything else. I have a BT100 running 1.0.6.7
code. When I go to the status page it says it's not registered
(hmm, that's not good). I also can't get dial tone but I can dial!
I'm afraid I'm lost any good
On Friday 05 Aug 2005 17:08, Neil Cherry wrote:
I'm seeing all sorts of problems and it's probably more of my lack
of experience than anything else. I have a BT100 running 1.0.6.7
code. When I go to the status page it says it's not registered
(hmm, that's not good). I also can't get dial tone
I have the Digium S100i IAXy device hooked up to my asterisk server.
When I pick
up the phone I do get dial tone but it does not stop when I start to
dial a number. The
dial tone is alway heard and it does not make the call.
It does register with Asterisk
I can make a call to the IAXy device
I recently have purchased a new TE110P card, that provides a single T1/E1
port. I have installed it and everything works fine, except for the dial
tones. When I made a call from a SIP phone to a channel in the TE110P, I
receive no dial tone. When I receive a call in a SIP phone from a channel in
On Mon, Jan 17, 2005 at 09:52:45PM -0700, Joseph wrote:
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter no-password I can make a call but if I enter any number
instead of word passcode it will
, but this one
works for me.
B. J.
-Original Message-
From: Paul Fielding [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 20:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] internal dial tone on password from outside
When I
Michael Greb wrote:
On Mon, Jan 17, 2005 at 09:52:45PM -0700, Joseph wrote:
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter no-password I can make a call but if I enter any number
instead of word passcode it will
Is it possible to get an internal dial tone when I call to my asterisk
and enter password?
I would like to call my line enter extension - password - and get
internal dial tone.
once I'm in I would like to dial based on what context permits, mostly
long distance calls VOIP.
I can not preset the
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
-
From: Brian Dingman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 7:43 PM
Subject: Re: [Asterisk-Users] internal dial tone on password from outside
http://www.voip-info.org/tiki-index.php?page
On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
Thank you!
DISA (Direct Inward System Access) - that is what I need.
DISA,passcode|context
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
Can someone
Joseph wrote:
On Mon, 2005-01-17 at 21:43 -0500, Brian Dingman wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
Thank you!
DISA (Direct Inward System Access) - that is what I need.
DISA,passcode|context
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
exten = s,1,Authenticate(X)
exten = s,2,DISA,no-password|local
Can someone explain to me what passcode is used for?
If I enter no-password I can make a call but if I enter any number
instead of word passcode it will not let me IN.
Is passcode a second level password; the
Hello all,
My problem is that when I call from an extension to another, I ear the
dial tones, but when I make a call using the Zap or Capi channels I do
not ear the dial tones.
Why this could be happen?
Any clue will be appreciated.
Thanks.
Ismael gil.
Hi all,
I have a Grandstream Budgetone 102 setup on a Asterisk PBX along with
an IAX adaptor. Everything seems to work fine. However, I would like
the dial tone signal to be generated from Asterisk when Budgetone is
picked up. It generates its own signal and does not really obtain it
from the
Shekhar Prasad wrote:
Hi all,
I have a Grandstream Budgetone 102 setup on a Asterisk PBX along with
an IAX adaptor. Everything seems to work fine. However, I would like
the dial tone signal to be generated from Asterisk when Budgetone is
picked up. It generates its own signal and does not
On Wed, Nov 17, 2004 at 10:27:38AM +1100, Duane spake thusly:
Shekhar Prasad wrote:
an IAX adaptor. Everything seems to work fine. However, I would like
the dial tone signal to be generated from Asterisk when Budgetone is
picked up. It generates its own signal and does not really obtain it
Tracy R Reed wrote:
nice feature. My clients always pick up the phone at first and assume that
because they have dialtone they had a good connection. Wouldn't some sort
of setting have to be changed in the phone to make the phone not generate
dialtone and wouldn't it have to be set to
nice feature. My clients always pick up the phone at first and assume that
because they have dialtone they had a good connection. Wouldn't some sort
of setting have to be changed in the phone to make the phone not generate
dialtone and wouldn't it have to be set to automatically connect to *
HengWee Chin wrote:
Hi,
I am having problem with the fxs port. I have compiled the zaptel,
zapata and asterisk version 1.0.1. But after I start asterisk, I do not
hear any dial tone coming on the fxs port. I am not able to dial out too.
Perhaps you could post your zaptel.conf and zapata.conf
Hi,
I am having problem with the fxs port. I have compiled the zaptel, zapata
and asterisk version 1.0.1. But after I start asterisk, I do not hear any
dial tone coming on the fxs port. I am not able to dial out too. But the
funny thing is that I am able dial to the fxs port. The phone ring in
HengWee Chin wrote:
Hi,
I am having problem with the fxs port. I have compiled the zaptel,
zapata and asterisk version 1.0.1. But after I start asterisk, I do not
hear any dial tone coming on the fxs port. I am not able to dial out
too.
Perhaps you could post your zaptel.conf and zapata.conf
Hi,
I'm trying to dial out on a vonage line connected to a zap channel
using stuff like:
exten = 200,1,Dial(Zap/2/${EXTEN})
but it doesn't work - when i dial in the extension, i can see on a phone
connected to the same line that it's gone active - but there's no
dialtone. also tried adding
]
[mailto:[EMAIL PROTECTED] On Behalf Of
Imran Akbar
Sent: Thursday, September 02, 2004 7:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] no dial tone when dialing out on vonage
Hi,
I'm trying to dial out on a vonage line connected
: Tuesday, June 22, 2004 9:49 PMTo:
AsteriskSubject: [Asterisk-Users] No dial tone after
installation
I'm a newbie,
after installing asterisk software and the Digium Lite card I do not get a
dial tone. The telephone push buttons makes tones but no dial
tone. Is this a hardware problem
Moskaluk
Sent: Tuesday, June 22, 2004 9:49 PM
To: Asterisk
Subject: [Asterisk-Users] No dial tone after installation
I'm a newbie, after installing asterisk software and the
Digium Lite card I do not get a dial tone. The telephone push
buttons makes tones but no dial
://www.moskaluk.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Sent: Wednesday, June 23, 2004 9:59 PM
To: [EMAIL PROTECTED]
Subject: Re: FW: [Asterisk-Users] No dial tone after installation
I forgot to add noload = chan_alsa.so
On Wed
]
[mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling
Sent: Wednesday, June 23, 2004 9:59 PM
To: [EMAIL PROTECTED]
Subject: Re: FW: [Asterisk-Users] No dial tone after installation
I forgot to add noload = chan_alsa.so
On Wed, 2004-06-23 at 20:43, Eric Wieling wrote:
/etc/asterisk
I'm a newbie, after
installing asterisk software and the Digium Lite card I do not get a dial
tone. The telephone push buttons makes tones but no dial tone. Is
this a hardware problem ?
Can some one point
me in the right direction.
Sincerely,
Don Moskaluk
[EMAIL PROTECTED]
On May 27, 2004, at 11:01 PM, Aaron J. Angel wrote:
Michael George wrote:
But, this isn't a big deal, we can live without it. I just
thought there might be a way. If I could do a
Backtround(Playtone()), that would do what I want...
There's no need for that. The playtone application continues to
Michael George [EMAIL PROTECTED] wrote:
I get the 9 and start PlayTones().
I go to the next context (with the tones playing).
In the next context (tones still playing) my matches are all several
digits long, so the tone is playing as the digits are pressed. That is
disorienting because
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I don't want asterisk to
mimic that
Michael George [EMAIL PROTECTED] wrote:
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I
On Fri, 28 May 2004, Michael George wrote:
Yes, I see what you are saying. And I tried this. Here's what happens:
I get the 9 and start PlayTones().
I go to the next context (with the tones playing).
In the next context (tones still playing) my matches are all several
digits long, so
It's true, if you're not careful, you could give incoming callers access
to your outside lines. But it is possible, with careful use of contexts,
to ensure that callers coming in on the context you specify for incoming
calls does not have access to the context that contains the dialplan for
Michael George [EMAIL PROTECTED] wrote:
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our current PBX which will just connect us to an outgoing
line (with a dialtone) when we hit 9. (Though I
The way I have my dialplan configured, an internal extension is routed
to a different context (with Goto()) on pretty much the first button
press.
2 - internal extensions
0 - operator
5 - VM
9 - outside line
etc.
So a 201 will go to the internal extensions context, s,1, do some
setup and then
ignorepat = 9
- Original Message -
From: Michael George [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, May 27, 2004 11:53 AM
Subject: [Asterisk-Users] generate dial tone
The way I have my dialplan configured, an internal extension is routed
to a different context (with Goto
: [Asterisk-Users] generate dial tone
The way I have my dialplan configured, an internal extension
is routed
to a different context (with Goto()) on pretty much the first button
press.
2 - internal extensions
0 - operator
5 - VM
9 - outside line
etc.
So a 201 will go
Michael George [EMAIL PROTECTED] wrote:
The way I have my dialplan configured, an internal extension is routed
to a different context (with Goto()) on pretty much the first button
press.
2 - internal extensions
0 - operator
5 - VM
9 - outside line
etc.
So a 201 will go to the
(Playtone()), that
would do what I want...
Thanks for the suggestions!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael George
Sent: Thursday, May 27, 2004 11:54 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] generate dial tone
The way I have my
ignorepat = 9
Ahhh... see that is something I did not grasp the concept of until now -- what
'ignorepat' actually did. Now I know that it defines the pattern of leading
digits received that Asterisk will NOT stop playing dialtone upon receiving.
I wonder if the documentation's been updated
On Fri, 2004-05-28 at 05:37, Michael George wrote:
On May 27, 2004, at 2:01 PM, Rechenberg, Andrew wrote:
I believe it's the 'ingnorepat' option that you want. Look at the
stock
extensions.conf and search for ignorepat.
I've tried ignorepat = 9, but that only seems to work within a
Michael George wrote:
But, this isn't a big deal, we can live without it. I just
thought there might be a way. If I could do a
Backtround(Playtone()), that would do what I want...
There's no need for that. The playtone application continues to the next
priority as it plays the tone, and
I just got the X100P and the TDM400P with one module on it, I had
installed asterisk and confirgured some file, but I can't get a dial
tone on my analog phone.
can someone help?
Regards
Leo
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hello List,
I just started messing with the settings on a SPA-2000, and it has a really
nasty alternative dial tone that I want to make go away. I'm not too hip on
how the two numbers interact, so my results haven't been good.
(I'm in the US, so I'm bias'ed towards US tones.)
Default(I'm ok
i've already tried to change the indications.conf to the following:
dial = 0/1500
but the dial tone still persists
i am using the following workaround but obviously not a clean
b'cos it just replace dial tone with some other tone.
in zapata.conf
context=spec
immediate=yes
Hello Everyone -
Well, I think I'm getting closer with the asterisk connection. This is my
setup and I keep getting this error below in ,my /var/log/asterisk/messages
file. I have opened 5060 port on the firewall box.
I would this is Warning which I can ignore! But I see the connetcion
You haven't quite supplied enough data to solve this problem.
Have you successfully used your ATA-186 on Asterisk when they're on
the same network segment (no firewall)? Is your firewall a NAT? It
appears that there is a NAT at both ends of this session. That
probably won't work, if that's
On Thu, 2003-04-03 at 09:58, Drew Hamilton wrote:
Any way to get the Asterisk to do the broken dial tone on phones that
have voice mail in their mailbox? Similar to the phone company's
built-in voice mail
If you are on a Zap device place a mailbox=box number before the
channel= line so
Yah, just set mailbox=foo above the channel declaration.
Mark
On Thu, 3 Apr 2003, Drew Hamilton wrote:
Any way to get the Asterisk to do the broken dial tone on phones that
have voice mail in their mailbox? Similar to the phone company's
built-in voice mail
- awh
On Thursday 03 April 2003 09:58 am, Drew Hamilton wrote:
Any way to get the Asterisk to do the broken dial tone on phones
that have voice mail in their mailbox? Similar to the phone
company's built-in voice mail
/etc/asterisk/zapata.conf:
;
; Stutter dialtone support: If a mailbox is
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