Displaying 20 results from an estimated 4000 matches similar to: "Authentication"
2003 Jul 03
3
Using switch =>
hello,
I have a test setup with 2 asterisk servers, each having a one snom 100
via sip using it. I`m experimenting on how trunking between them would
work. I have them setup for RSA authentication which I plan to use in
the future.
So I`ve setup the keys and servers seem authenticate to each other. One
is named phila and other hurricane.
Here is what I see on phila:
-- Registered
2003 Dec 15
2
Beginner couple of questions
Dear all,
I have some questions, I'm sure it's pretty stupid for most of you, but I need
you guys to help me. Here are my questions:
1. Music On Hold, it doesn't play any sound on the parked call or hold call.
But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using
slackware 9.1
2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,
2004 Aug 04
10
htb and fw problems
Dear All,
I''m using the kernel 2.6.6, iproute2-2.4.7.20020116, iptables v1.2.9, and gentoo.
I have a leased-line 64 kbps.
I can see the counter works in iptables, but in the htb, it doesn''t go to the right class (it always go to the default class).
Any help will be appreciated
here''s my htb conf
#!/bin/bash
tc qdisc del dev eth1 root
tc qdisc add dev eth1 root
2004 Jan 15
3
Sending voicemail with qmail
you can do that. But are u installing qmail and * on
same box. i wont
recommend that. i use qmail and *. qmail is strictly
for internet email. *
is on separate server not exposed to Internet. * box
also has sendmail. i hv
configured sendmail to use smart host (qmail server).
This way its safe and
secure.
HTH,
-B
----- Original Message -----
From: "Ing Isianto Istiadi"
2006 Oct 13
2
AEL Question
Hi, all. I'm making my first foray into AEL. I'm starting with a
simple macro, but I've already encountered an odd behaviour. I'm
wondering if someone can shed some insight:
Asterisk 1.2.9.1
macro newPlaceCallPSTN {
s => {
TIMEOUT(absolute)=7200;
NoOp(Analog Out List: ${ANALOGOUT});
ChanIsAvail(${ANALOGOUT});
NoOp(Available Out List:
2003 Sep 18
4
New message 0 in mailbox 7606
Hello,
I recently started playing with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives.
1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this?
2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the
2003 Aug 07
1
MWI bug ?
Hi Lee,
You need to specify the VM context that you are using..
so using your examples..
extensions.conf entry..
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000)
exten => 1000,102,Voicemail2(b1000)
exten => 1000,103,Hangup
should be..
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000@sip)
exten => 1000,102,Voicemail2(b1000@sip)
exten
2003 May 10
19
Voicemail2
Asterisk Users:
I've been working hard on app_voicemail2 which is an enhanced scalability
version of app_voicemail. Specifically, its features are:
* Highly improved internal architecture (maybe someone else can
actually code on it)
* Foot print for getting mailboxes from DB (for Vonage)
* Segmentable mailboxes, allowing you to truly multihost
voicemail for multiple companies
2004 Sep 23
1
running 1.0 on macosx
Hi,
compiled 1.0 on macosx latest (10.3.5). compiled fine. when running,
complains about voicemail2 module. Any hints?
Marc.
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-HEAD-09/23/04-09:20:48, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <markster@digium.com>
2003 Jul 16
4
voicemail instructions
Hi,
I've been playing with Voicemail and Voicemail2 a bit for my users, and
there are a few things I'm wondering about:
- We can specify parameters to the mailbox (s, b or u) to select which
prompts to play. However, if we specify 'b' or 'u' it plays that
(customisable) message, but it also plays the voicemail instructions. For
the dutch, it is customary that a user
2003 Sep 16
8
Hangups after voicemail
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung up.
Therefore, I've added the following hideous hack - er, code - to voicemail2.c.
It
2005 Mar 11
3
Parked Call
I have a question,
I am unclear on how to park a call. I know that you are supposed to be
able to press "#" and then transfer the call to extension 700. However,
* doesn't seem to be graping the dtmf. I am using dtmfmode=inband.
Asterisk is in the media path as well.
Thanks in advance
Justin
2003 Nov 06
2
Voicemail2 vs Voicemail
>> Wouldn't that break everybody's dialplans where they would have to
>> replace all occurrences of Voicemail2 with Voicemail and all
>> occurrences of Voicemailmain2 with Voicemailmain?
>
> No, we would register with both names.
Is it necessary (with reasonably current cvs) to make any changes in the
*.conf files to use Voicemail2, or is that happening
2003 Oct 27
3
passing digits for voicemail from sip gateway
I am seeing strange behavior that I don't understand. Voicemail2 and
voicemailmain2 work fine if I call from a sip phone directly connected
to *, but if I call either of them from an analog line on the other side
of a sip gateway, voicemail seems to ignore digits. If I am recording a
message and press #, nothing happens except that it records the tone
onto the message and I can't specify
2003 Oct 25
1
Voicemail.conf in MySQL is not functioning
Voicemail.conf in MySQL is not functioning where I get the following error
from Asterisk messages log file:
CLI debug output is as follows:
Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack
-- Playing 'vm-login'
-- Playing 'vm-password'
-- Incorrect password '1234' for user '0' (context = <any>)
-- Playing
2003 Aug 08
4
Voicemail2 - auto fill the dialing extension?
Hi,
First off, a big thanks to Digium (Mark, John, and Martin) for helping
sort out a BellSouth config issue on our PRI. T100P working like a
champ!
Now it's back to tweaking the configuration on our SIP phones (7960s).
The message_uri parameter in the phone's configuration file is working
great. Dials comedian mail directly. Is there a way to let voicemail2
know what the incoming
2004 Jul 08
2
HOW ASTERISK WORKS
Hy guys, I cannot understand How the asterisk works. I
would like know how the h323.conf, sip.conf and
extension.conf works. I don't understand the
parameters and the [sections].
What I need to the asterisk get a SIP call and forward
them to a H323 terminal. I working at the h323.conf
and extension.conf but I cannot understand!!! Please
someone can help me.
I your can send me a example (with
2003 Sep 22
1
Voicemailmain2 user docs?
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user documentation hiding
in Internet land some place? If not there well
be soon. Ho hum.
2003 Aug 18
3
Voicemail2 vs. Voicemail
Does anybody have any reason why I should *not* permenantly replace
app_voicemail with app_voicemail2? If so, speak now or forever cvs update
-D "8/18/2003".
Mark
2004 Jan 05
8
This newbie gives up for now - sadly
This newbie has been trying out Asterisk. It has been both a) surprisingly
painful and b) impressive in terms of helpful support from other users.
Having got two phones to communicate and then got voicemail MWI going
(neither painlessly) I decided the next step was to implement call transfer
as per nearly all commercial PBX systems i.e.
hold call
consult another extension
either exit and let