Hi!
My application needs to look up (by spelling) the first and last names of a
person and then insert the corresponding pre-recorded audio file to
personalize the message. E.g. Hi, John Brown. Your book is due back at the
library. So I need John and Brown in audio files along with LOTS of
other
Title: Very decent book - VoIP Telephony with Asterisk
Just received my copy of this book today. Based on a cursory examination, VoIP Telephony with Asterisk (VTwA) by Paul Mahler looks like a very decent book for anyone starting out with Asterisk. More knowledgeable readers may also
in the book. No need to flame me
on that fact. I am a regular consumer like anyone. Author felt
inclined to put it in there.
- Original Message -
From: John Vogel [EMAIL PROTECTED]
Date: Fri, 23 Jul 2004 22:44:35 -0700
Subject: [Asterisk-Users] Asterisk for Small Office Setup
Title: Do not buy Asterisk for Small Office Setup!
See my previous email but the book is worse than I thought. In addition, to the things I mentioned earlier.
1. The table of contents is not a table of contents. The only chapter heading is Chapter 1. It is impossible to tell what is in
putting it up for sale? :)
Thanks!
Remco
On Sat, 24 Jul 2004, John Vogel wrote:
See my previous email but the book is worse than I thought. In
addition, to the things I mentioned earlier.
1. The table of contents is not a table of contents. The only chapter
heading is Chapter 1
Title: Asterisk for Small Office Setup
Don't buy this book for its content. It's a waste of $40. However, it is useful to wave in front of my customer's faces to show them that Asterisk is real.
Yeah, but how do you get one? We can't find a single dealer with any in
stock -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James H.
Thompson
Sent: Thursday, July 15, 2004 2:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP phones
Title: Asterisk and dial-up modems
Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other?
Title: Modems behind Asterisk - how?
The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main
Title: Call centers using Asterisk
I have a local company that wants to use Asterisk for a small call center. I've told them that they can do so but they'd like to actually talk to somebody that is. Does anybody know anyone using Asterisk in a call center that they could talk to?
If you're
Title: Unsupported Media error from iConnectHere
I can't talk through iConnectHere. The connection gets made but as soon as any sound is transmitted the call ends and the Asterisk console shows an Unsupported Media error as follow:
Got SIP response 415 Unsupported Media back from
Title: iConnectHere broken?
I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a Unsupported Media error even though I'm still using ulaw and alaw.
It stopped
in Portland
John Vogel wrote:
Anyone else going to this and interested in having an Asterisk
birds-of-a-feather breakfast or dinner get together?
Open Source Convention,
Portland, OR,
I am, and would be very interested.
Did you notice that John Todd is doing an Emerging Topics session
Title: SwissVoice IP phones
Has anybody used these with Asterisk?
Thanks to all for the comments even if they don't agree!
I think this issue is significant and I would really like it to be fixed in
the 1.0 release.
Does anybody know how to get the same functionality without using *8?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Title: O'Reilly Open Source Convention in Portland
Anyone else going to this and interested in having an Asterisk birds-of-a-feather breakfast or dinner get together?
Open Source Convention,
Portland, OR,
July 26-30.
See http://conferences.oreilly.com/oscon
: Re: [Asterisk-Users] *8 problem still there?
I see this (but not using a recent asterisk version)
I had put it down to a software bug in the grandstream phones that I'm using
- are you sure its an asterisk bug or are you using grandstream also?
Steve
On Mon, 17 May 2004, John Vogel wrote
Title: Vertical applications?
Has anyone created any vertical applications, e.g. real estate, for Asterisk?
I'm trying to market * in my area (Seattle) and would like to offer vertical apps to my customers. These apps will help me compete with the big guys like Cisco, Avaya, etc.
If you
Title: Some doc
I've posted a couple of docs that are intended to be helpful at http://www.rainiernetworks.com/29535/33701.html
1. A case study of one of our customers using a fairly straightforward T1 coming in to the Asterisk box
2. The installation steps we followed
3. The conf
Title: *8 (call pickup) using Manager or AGI interfaces?
I'd like to programmatically do the equivalent of a *8 using either the Manager or AGI interfaces or some other Asterisk interface.
In this scenario, a line is ringing, the pickup program executes with an argument that is the
For under a $75 I like
http://www.bizrate.com/marketplace/product_info/details__cat_id--1167,pr
od_id--7582541.html
It's the Panasonic KX-TSC14B Phone two-line, w/caller ID, speakerphone and
inexpensive ($60). Various other features, too.
-Original Message-
From: [EMAIL
All:
Why do some requests to the manager return [--END COMMAND--] and some don't?
(version 0.7.1)
In the following example show version has it and sip show peers doesn't.
Why?
Thanks for any suggestions!
[Action: Command, Command: sip show peers]
[Response: Follows]
[Name/usernameHost
Four or five analog lines can be done with a single computer so no channel
bank is needed. If you need 6 or more than there is also the choice of using
two machines and IAX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Hajime
Lanning
Sent:
Doesn't work for me. Connects to Asterisk but says All extensions are busy
right now when I try to do anything. Here's what an extension looks like.
Any suggestions? Thanks!
Extension
NameExt 2003/Name
Number2003/Number
DeviceSIP/2003/Device
Contextfrom-sip/Context
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgi
Or did you mean asynchronously?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
Lewis
Sent: Friday, April 02, 2004 6:27 AM
To: [EMAIL PROTECTED]
Subject:
If price is an issue but you have the slots for 3 - 4 cards you could try
DigitNetworks. Their X100P compatible cards are only $28.50 (US).
My quick evaluation of the alternatives is:
1. Adtran 750 channel bank or something similar. This can handle up to 24
FXO lines and converts them to T1 to
What version of *? I'm using 0.7.1 and it still has occasional problems
detecting call hangup.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, March 31, 2004 8:54 AM
To: [EMAIL PROTECTED]
Subject: Re:
Do you have
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
In your zapata.conf file? Wiki is good for this -
John V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Justin Carlson
Sent: Thursday, April 01, 2004 9:45 AM
To: [EMAIL
on both the box with the zap interface and the remote office. it helped
some but the problem remains
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Vogel
Sent: Thursday, April 01, 2004 12:10 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Echo's
I am still experiencing the problem where you pick up an incoming analog
call ringing on SIP Phone A with SIP Phone B using *8 but Phone A continues
to ring. This happens with Grandstreams and Snoms on the 0.7.1 code base.
My theory is that Asterisk is not telling Phone A to stop ringing when
If you ever get an answer to this please let me know off-line,
[EMAIL PROTECTED]
I have a security expert friend using Asterisk who is interested in running
a whole set of such tests on it. My theory is it is security swiss cheese.
Thanks, John V.
-Original Message-
From: [EMAIL
Title: Call pickup - still keeps ringing?
On my system with 0.7.1 call pickup from SIP to SIP still leaves the originally dialed phone ringing for 10's of seconds after the call has been picked up on another line.
There was a post a back in the fall that said this had been broken in a code
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