Displaying 20 results from an estimated 8000 matches similar to: "Help! No stutter dialtone on message waiting - zaptel phones"
2004 Jan 30
3
How do you turn on the 7960 msg waiting light?
Does anyone in Asterisk land know how to turn on the message light on the
back of the earpiece of a cisco 7960 when a message is waiting?
Thanks!
Paul
Paul Mahler
mail:pmahler@signate.com
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2006 Apr 08
1
unable to enable stutter dialtone
I'm having problems enabling stutter dialtone for users connected to
channel banks.
Half of our users are on iaxy's and the other half are connecting to
channel banks. The users on ixay's are getting the stutter dialtone
on new voicemails, but the ones on the channel banks are not.
Currently, all users are in the default context in the voicemail.conf
file. I've tried the
2004 May 19
1
Using stutter dialtone like the PSTN does
A question: is there any way to get * to answer certain DTMF sequences
entered on an extension with a stutter tone?
Long version: I would like to add features to my dialplan like "Caller ID
Unblock" which work in the same way that the PSTN works: I pick up
the phone, get a regular dialtone, press *82, and get a short stutter
dialtone which confirms acceptance of the request, and then
2003 Dec 18
2
Cisco 7960 - can't traverse NAT?
Might be a stupid question, but is there a default gateway set on the 7960?
-----Original Message-----
From: Paul Mahler [mailto:pmahler@signate.com]
Sent: Thursday, December 18, 2003 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT?
I have a 7960 running behind a firewall running NAT. From a telnet session
to the 7960, I can't ping
2004 May 14
7
What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten => 99,1,VoicemailMain(${EXTEN}@inside)
exten => 99,2,Hangup
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2004 Mar 17
4
can't logon to voice mail - bad password
I have one SIP extension that can't logon to voicemail. The log file says
-- Incorrect password '3213' for user '4035' (context=other)
even though the context in voicemail.cnf says
4035 => 3213,Bill Smith
Thanks!
Paul Mahler
mail:pmahler@signate.com
phone: 650.207.9855
fax: 877.408.0105
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2004 May 14
4
How to Echo extension number to caller?
I need to dial an extension that tells me what extension I'm dialing from.
I'm running a bunch of analog phones off a channel bank to * over a T1. I
have the following in extensions.conf.
exten => 98,1,SayDigits(${EXTEN})
This says the digits the caller enters on the keypad, not the extension they
are calling from.
Thanks Guys!!!!!!!!
Paul
Paul Mahler
pmahler@signate.com
2006 Jan 09
1
Second edition of my * book has been released
How does it compare with the O'Rielly book?
Does it include information on CVS, or primarily on stable?
Can it be provided to customers, or is it more sysadmin oriented?
Regards,
Greg
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul
Mahler
Sent: Thursday, January 05, 2006 9:45 AM
To: 'Asterisk
2004 May 10
1
Terrible TICKING sound
i'm getting a tick every second or so on all my calls. All channels are zap
channels.
Does anyone know how to fix this?
Thanks!
Paul
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2004 May 02
1
Why don't I get a ringing sound?
I am using the following macro to dial a ZAP channel. When I dial in, *
answers and I go to voicemail. I never hear any ringing, though. It doesn't
work with the Ringing command before or after the Dial command.
[macro-zapdial]
;
; call a ZAP extension for ${ARG2} seconds, and then voice mail
; ${ARG1} - Extension
; ${ARG2} - Time to ring
exten => s,1,Dial(ZAP/${ARG1},${ARG2})
exten
2003 Jun 04
1
new application Dialtone()
Hello,
I created a new application for myself called Dialtone() by modifing
res/res_indications.c file. It can be used as such:
exten => s,4,Dialtone(30|${CALLERIDNUM})
exten => s,5,Playback(time-exceeded)
exten => s,6,Goto(s|1)
It will stutter if you have new voicemail and you have passed the mailbox
number as I did above. It will stop dialtone the moment you press a key
2004 May 24
3
100 analog phones?? HOWTO?
Does anyone know the best approach to take for handling 100 analog
phones? It seems to me that a chassis like Carrier Access or Adtran
would work. The chassis would do much of the hard work of converting
the analog sound to data.
Any recommendations on hardware for the chassis?
...Jeff
2004 Jun 02
0
Stutter dialtone on TDM31B (TDM400P)
I think I've configured everything to have stuttered dialtone on my
analog phones (it works fine for my SIP phones). But I still don't have
it:-( I'm using asterisk 0.9.0 and zaptel 0.9.1.
In voicemail.conf:
[local]
21111 => 987654,Robert Withrow,email@here.com
In zapata.conf
[channels]
...
mailbox=21111@local
channel => 1
2004 Nov 19
5
Asterisk and H.323 Gatekeeper
Hello,
I am new to this list and to asterisk and going through the archive file I
did not find an answer to my problem.
I have a VoIP network working fine with multiple gateways registered to a
Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in
that network and also successfully registered two X-Lite SIP Client to
asterisk that call to each other.
I want to connect to
2003 May 25
2
Message Waiting and VoiceMail 2
Hi.
I noticed that if new messages are recorded
with voicemail2 , they're not detected by
the message waiting indicator, so
the mailbox=XXXX param has no effect, and
no message waiting is sent to the phone
(sip & adsi, or stutter dialtone)
Any hint?
--
Brancaleoni Matteo <mbrancaleoni@espia.it>
Espia - Emmgi Srl
2004 Jul 10
5
Three (quick?) questions...
[Please excuse if this is a repeat; I initially tried to send it from a
different account, and it's been held up for a couple of days awaiting
moderation.]
1) What's the absolute minimum required (hardware-wise) in order to get one
in-bound POTS line into Asterisk, and then have IP phones "inside?"
[In other words, I obviously need a NIC -- but what would be the
2006 Jan 12
0
Second edition of my * book has been release d
But for us?
_____
From: William Boehlke [mailto:william.boehlke@signate.com]
Sent: Wednesday, January 11, 2006 2:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Second edition of my * book has been released
$39.95 retail.
_____
From: asterisk-users-bounces@lists.digium.com
2005 Mar 04
7
Stutter Tone
I think I have something misconfigured regarding voicemails. They work
great, I have this setup:
Sip.conf
[ext1]
Context=phones
Mailbox=201
Voicemail.conf
[home]
201,password,name,email@mail
Voicemail delivery and all works great but when I check sip extension ext1
(analog phone using a Granstream ATA 286), the stutter tone signaling
message waiting does not work.
Anything wrong with
2005 May 18
2
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2004 Jun 18
0
SIP error 407 - can't make outgoing calls
I am using a IPDialog siptone II. I can take incoming calls, but when I try
and make an outgoing call I get a SIP 407 error.
Can some kind soul explain to me what I am doing wrong?
Here's what I found in the wiki:
If a proxy does not accept the credentials sent with a request, it SHOULD
return a 407 (Proxy Authentication Required). The response MUST include a
Proxy-Authenticate header