Displaying 20 results from an estimated 1100 matches similar to: "HOW ASTERISK WORKS"
2004 Jun 24
1
ZyXEL Prestige 2000W and DTMF
I've just seen this post:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html
and it took me back to play again with my dust collecting 2000W. Does
anybody got DTMF to work?
My sip.conf looks like this:
[400]
type=friend
context=from-sip
username=400
secret=verysecret
disallow=all
allow=g729
dtmfmode=rfc2833
host=dynamic
nat=yes
qualify=300
canreinvite=no
My phone is
2004 Sep 21
4
Voicemail forward to a remote server?
Anybody ever managed to implement a solution where one could forward a
voicemail from one * server to another?
Dominique
2004 Jul 13
5
WiSIP and Zyxel Prestige 2000W
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Hash: SHA1
Hi,
Anyone have any experience with either of these, I 'd appreciate some
feedback? Plus it seems pretty easy to steal a connection with this.
Zyxel Prestige 2000W
WiSIP
thanks,
- --
Steve
"They that would give up essential liberty for temporary safety deserve
neither liberty nor safety."
Benjamin
2004 Aug 10
2
WiFi phone radiation regulation?
All,
I just had the fortune to take one of the new Senao Wifi SIP phones for
a short test drive. First look - it's a nice, compact phone. Weighs
around 87g and roughly the size of a Nokia 6210. More on the those
later. The thing that struck me was the RF power, it's rated at 100mw
(20dBm). That's 10 times more than any of the other brands out on the
market Cisco, WiSIP, Zyxel
2004 Jun 24
2
R: R: R: How to force G729
> "If" I understood your initial objective correctly (and I may not have),
> the user's phones are negotiating the codec to be used for each rtp session.
>
> Asterisk parameters can be used to dictate rtp sessions between the sip
> phone and asterisk, but that won't influence the next step in which the sip
> phone negotiates a new rtp session directly with the
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 Jun 26
2
ZyXEL Prestige 200w - should I return it ?
Hi all
I have just got a P2000w and experience several problems. Hopefully there is
someone out there that has got it working. I saw it on Cebit and the person
demonstrating it there told me that it was connected to an Asterisk server
on the stand -so it should work.
Problem 1: it does not register correctly
It get lots of messages like this:
Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630
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"
2005 Jun 29
2
Asterisk LAMP Developer
_Description_
We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP)
developer with some Asterisk experience who is based in Western or
Eastern Europe or Asia. We can work with an individual or an
organization. You must be fluent in English.
We need you to help expand development of Signate's core software products.
As part of the Signate development team, you will
2004 Jun 02
5
ZyXEL Prestige 2000W SIP hangup fails
Does anybody have any experience with the ZyXEL Prestige 2000W? I am
having problems with the line tear down when I call another extension.
If nobody picks up at the other end when I hangup the 2000W, the other
extension continues to ring. Is there any way to hangup a SIP call if
there is no more traffic? Asterisk seems to think that there is still a
connection open. This is pretty annoying
2003 Dec 22
1
Authentication
Dear all,
I have a question regarding the configuration of *. I have 3 port FXS, and 2
port FXO. I have 4 users that use analog phone connected to FXS (I have 3
phones). I need to limit the user's capability (user A can call
International, user B can call long distance, etc). I want to implement the
password say to call , he/she needs to puch 9(for outgoing call)2-4 digits
password,then
2004 Jul 07
0
Conf files doubt
Hello guys, I am here again, sorry for borring but I
in freeze here! : )
Just for resuming some doubts:
+ The extension.conf file:
has a [general] context for "general" configurations,
and a [global] context for global variables. The
another context [any_name] are context for "handle the
calls",
for example:
exten => 1001,1,Dial(SIP/1001@10.11.2.121,30) means,
when the
2004 Jul 12
0
GnuGK + Asterisk + SIP Provider
Hi guys, I create a topology like fellow:
/****** /************ /***********
* GK *-----------* Asterisk *---------- Sip Prov *
******/ ************/
***********/
| |
| |
| |
H.323
2004 Jul 12
0
GnuGK + SIP Provider + Asterisk
hi, just change the problem in another email. I can
make calls from H323 to Sip, but cannot on the other
hand, i.e. from SIP to H323. I am thninking that the
asterisk don't response the REQUEST of the Sip
Provider. Or, the GNUGK don't do a response to the
asterisk that don't malke a response to the Sip
Provider!!!!
AnyIdea!?!? Thanks in advance.
Giscard
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
2004 Jun 18
2
C7960 g729 question
I have multiple voiceage g729 licenses installed on a RH9 box, and have
a remote C7960 configured to use it (low bandwidth). In calls like:
Remote C7960 -> g729 -> asterisk -> g711 -> C7960
the audio is oftentimes rather choppy. Changing the remote 7960 to use
g711 seems to eliminate/reduce the choppyness. Any ideas on what might
be behind this?
2003 Dec 15
6
more questions
> 3. Supposed I have 2 fxo cards (right now I have one already) and 3
> fxs, and one of the fxo will have two phone (running pararell), is
> there any way for * to:
> a. It always dial the first fxo, if the fxo is busy or is being used
> (have other people conversation), will * be able to switch it to
> other fxo? Here's the approximiate the conditions of the phone.
2010 Nov 23
2
"help"
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Nom : non disponible
URL : <https://stat.ethz.ch/pipermail/r-help/attachments/20101123/db21b78a/attachment.pl>
2004 Jul 14
3
Voicemail/autoattendant not working
I'm pretty much a newbie to this but still think I've been around the
various help pages, voip-info.org etc to be fairly sure I'm not
missing something here so your help is appreciated!
I have a box running RedHat9 at home with the latest CVS of Asterisk
and all works fine.
At the office, we installed Gentoo linux on a machine, downloaded the
latest CVS of Asterisk, set it up. All