I will also need to get an incoming number, which is more money, before I'm
satisfied with outgoing calls
But thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson
Sent: Thursday, November 06, 2003 4:08 AM
To: [EMAIL PROTECTED]
Great. And if you don't mind us guys who'll use it to prove asterisk's
working to management, it's even better.
Any chance on adding a dns capable script?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Thursday, November 06, 2003 7:49
Olle.
I've been in the mailing list for a couple of weeks now.
Many threads are answered with links to your wiki.
Cause of the DNS problem I can't get there, no matter what.
Till this is resolved, are you able to provide me (and many others) the
legit IP address for the web server for me to put
] On Behalf Of Olle E.
Johansson
Sent: Thursday, November 06, 2003 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Shoval Tom wrote:
Olle.
I've been in the mailing list for a couple of weeks now.
Many threads are answered with links to your wiki.
Cause
] archives gsm of asterisk ???
Shoval Tom wrote:
Setting it in hosts doesn't do me any good.
Trying to surf to http:// 64.65.102.50 gets me to apache test page.
Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
Get a 404 page doesn't exist.
Its most likely on a name based
You need to install some softphone, you can't interface asterisk to the
headset by itself.
Try x-lite from www.xten.com for windows (or Dan the man's DIAX software -
search the archives)
Or gnuphone for linux.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Shoval Tom wrote:
Setting it in hosts doesn't do me any good.
Trying to surf to http:// 64.65.102.50 gets me to apache test page.
Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
Get a 404 page
Dan the Man,
I was trying to use DIAX to call via iconnect (with asterisk in the middle)
It fails, for no apparent reason.
* consle show its trying to connect.
While x-lite continues on and bridges the two (x-lite to * and * to
iconnect)
DIAX doesn't, it just hangs up.
Any ideas?
-Original
of asterisk ???
Shoval Tom wrote:
Guys, it still not working.
Go here
http://www.checkdns.net/quickcheck.aspx?domain=voip-info.orgdetailed=1
And see that it returns errors.
PLEASE help.
None of the reported errors are critical.. They are just saying that
only one DNS server is active..
Try setting
It works with SIP and with zap channels.
What about IAX? like DIAX softphone?
I may be misunderstanding something.
When you start an Asterisk configuration process, connecting your hardware
and building your dialplan, you use Zapata.conf, sip.con and iax.conf to
connect the FXSs and FXOs to your
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Wednesday, November 05, 2003 3:45 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Hi,
- Original Message -
From: Shoval Tom [EMAIL PROTECTED]
To: [EMAIL PROTECTED
How is it not economical?
I already have the PBXs on both sides.
If I switch to * I'll need to get a channel bank
Am I wrong?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hkirrc.patrick
Sent: Wednesday, November 05, 2003 8:36 PM
To: [EMAIL PROTECTED]
As far as I can gather, the voicemailmain program is not configurable.
Please correct me if I'm wrong.
The other way to create a voice mail main of your own is to create a menu
with many submenus in extensions.conf - and that's no walk in the park.
-Original Message-
From: [EMAIL
I'm pretty much guessing,
But I suggest doing one of these
a. disable the modem module from loading (not sure how it's done)
b. run make samples from /usr/src/asterisk and use modem.conf that was
created. - that worked for me, but I guess is not the right solution.
Good luck
-Original
Jared, regarding your million minutes,
What is your internet connection (bandwidth, type, etc.)
Can someone gauge asterisk bandwidth consumption, and/or monitor it?
I think this is most crucial for a production system, as the world of VOIP
introduces us to a type of problem never encountered
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jose Quinteiro
Sent: Thursday, November 06, 2003 6:51 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk
Phone System?
You can't beat the simplicity and
Dan, problems discovered
First, DIAX still crashes after hitting exit, and not only from system tray.
Second, Why don't I get a busy signal? Even when I call myself. How many
lines(calls) is DIAX capable of having concurrently?
Third, I was playing with extensions.conf while adding users using
Me too!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Tuesday, November 04, 2003 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anyone using * in a live production
environment?
Gavin Hamill wrote:
Hullo again, all :)
If
First, DIAX still crashes after hitting exit, and not only from system
tray.
All the time or after some specific operations?
Did you registered with Asterisk server?
[Shoval Tomer] Every time I exit, and I have registered successfully.
Second, Why don't I get a busy signal? Even when I call
Dan, the software crashes if you exit after hanging up.
If exiting without doing anything first, it works OK.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Tuesday, November 04, 2003 5:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Actually asterisk.org has
all the info you need.
Just install the linux
distrib with CVS, kernel sources and openssl-devel and all their dependencies.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew England
Sent: Monday, November 03, 2003
3:56 AM
To:
Look into www.digium.com.
Digium's cards are you best choice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 4:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Questions
hello,
I am completely new to
Look into AGI, there a re some examples out there, but it's very much
doable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 11:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Another newbie question
Thanks
LinkSys NAT Routing
Shoval Tom wrote:
Isn't putting asterisk on the public IP network a bad idea?
Is it a bad idea?, Not really if you take the right precautions..From
how you described your setup you have connected your server directly to
the internet anyway.. If you nominated you Asterisk box
That is correct. I'm able to get to your site using the IP address provided
below.
Since I get the same address (192.168.168.3) from four different ISPs (home,
HQ, branch office, and dial-up to another one) I think it's safe to say your
DNS configuration is what should be looked at first.
Dan, any chance getting a look at the code?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peer Oliver
schmidt
Sent: Monday, November 03, 2003 8:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)
Will extern IP work if I had multiple phones connected behind NAT?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Monday, November 03, 2003 8:35 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
other places (like our branch office, my home dial-up account, my parents
dial-up account)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Monday, November
PROTECTED]
Subject: RE: [Asterisk-Users] Rollout tips
On Tue, 4 Nov 2003, Shoval Tom wrote:
Olle, www.voip-info.org still resolve to 192.168.168.3 from here, and many
other places (like our branch office, my home dial-up account, my parents
dial-up account)
Do you by any chance use the same
Either it's not working, or I don't know what I'm doing. It's giving me the
error sox: effect '.gsm' is no known!
Let's say I need to convert file 1.wav to 1.gsm.
How do I apply this command to it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel
So true, yet so irrelevant for my purposes.
I needed to convert existing IVR sound files to gsm, in order to demonstrate
asterisk's functionality to my bosses (the ones who'll pay for the hardware,
eventually...)
Besides, even if I didn't have the files ready, I wouldn't use my lovely
voice for
and it'll convert all *.wav files
for you.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Sunday, November 02, 2003 9:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] recording files for menues
Shoval Tom wrote:
Either
How should I configure Asterisk to allow this soft-phone to register?
Please provide an example
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rnc Info Lists
Sent: Sunday, November 02, 2003 11:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
Thanks for the detailed answer, and sorry about the not so detailed
question.
So here's my humble request.
Can someone who has implemented a live production Asterisk deployment,
preferably between two sites (HQ and a branch office, connected over the
internet) spare the time and contact me here,
too !
Now
works great!
Thanks
alot.
On Fri, 31 Oct 2003 19:12:11 +0300, Shoval Tom wrote:
I fixed it.
finally.
I was missing suidperl.
Apparently it isn't installed on
redhat 9.0
You can download it from
www.redhat.com and install it. And voile
everything works.
From:
[EMAIL
Thanks, but no go.
I already used these. And it still doesn't work.
Anything I can do about the horrible echo in x-lite?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Thursday, October 30, 2003 8:55 PM
To: [EMAIL PROTECTED]
Subject:
Well, found the answer for the DTMF problem, and
guys, the voicemail is G R E A T !!!
The answer was use rcf2833 for dtmfmode,
not inband as suggested earlier
If someone can help me resolve the cgi problem, I'd
be forever indebted
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
this probably is a newbie question, but
the voicemail web interface is a great selling point for the ones upstairs.
Thanks a lot for any answer.
Shoval
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tom
Sent: Friday, October 31, 2003
12:00 AM
To: [EMAIL PROTECTED]
Subject
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