From chris@tooley.com Wed Oct 12 07:38:01 2005
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Having heard no complaints about the idea of at least doing the first
meeting here, can I assume that's acceptable?

-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From lelliott5@gmail.com Wed Oct 12 08:01:03 2005
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Date: Wed, 12 Oct 2005 07:53:59 -0500
From: Les Elliott <lelliott5@gmail.com>
To: Chris Tooley <chris@tooley.com>
Subject: Re: [A*UG] Welcome, First Meeting
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Where is the office and when do you plan to meet?
Les

On 10/12/05, Chris Tooley <chris@tooley.com> wrote:
>
> Having heard no complaints about the idea of at least doing the first
> meeting here, can I assume that's acceptable?
>
> --
> Chris Tooley
> 512-646-1507
> ctooley@gnumber.com
>
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
>

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Where is the office and when do you plan to meet?<br>
Les<br><br><div><span class=3D"gmail_quote">On 10/12/05, <b class=3D"gmail_=
sendername">Chris Tooley</b> &lt;<a href=3D"mailto:chris@tooley.com">chris@=
tooley.com</a>&gt; wrote:</span><blockquote class=3D"gmail_quote" style=3D"=
border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddi=
ng-left: 1ex;">
Having heard no complaints about the idea of at least doing the first<br>me=
eting here, can I assume that's acceptable?<br><br>--<br>Chris Tooley<br>51=
2-646-1507<br><a href=3D"mailto:ctooley@gnumber.com">ctooley@gnumber.com</a=
>
<br><br>_______________________________________________<br>Austin-Asterisk-=
Users-Group mailing list<br><a href=3D"mailto:Austin-Asterisk-Users-Group@b=
ybent.com">Austin-Asterisk-Users-Group@bybent.com</a><br><a href=3D"http://=
buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group">
http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group</a><br>=
</blockquote></div><br>

------=_Part_1169_8390828.1129121639568--
From chris@tooley.com Wed Oct 12 08:22:32 2005
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On Wed, 2005-10-12 at 07:53 -0500, Les Elliott wrote:
> Where is the office

3925 W Braker Ln (The MCC Building at MoPac and Braker)

> and when do you plan to meet?

That was the next order of business, when can everyone get away to have
our first gathering.

> Les
> 
> On 10/12/05, Chris Tooley <chris@tooley.com> wrote:
>         Having heard no complaints about the idea of at least doing
>         the first
>         meeting here, can I assume that's acceptable?
>         
>         --
>         Chris Tooley
>         512-646-1507
>         ctooley@gnumber.com 
>         
>         _______________________________________________
>         Austin-Asterisk-Users-Group mailing list
>         Austin-Asterisk-Users-Group@bybent.com
>         http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 

From chris@tooley.com Wed Oct 12 08:30:24 2005
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> That was the next order of business, when can everyone get away to have
> our first gathering.

How is Monday the 17th for everyone?  Third Monday of the month
schedule?

From gwwalker@behemoth.gnumber.com Wed Oct 12 08:51:20 2005
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Does anyone know if my "NEW LIST CREATED" message made it out on ALG?
If not, would someone on the ALG list (and the GWLUG list) forward the
message?

I got the "moderator has your message" type message, but nothing saying
whether or not it went.

-- 

Wayne Walker

wwalker@gnumber.com                   Do you use Linux?!
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First real posting!

While another member offered his office space (a little larger than
ours) for meetings, Chris Tooley is by far the most knowlegable Asterisk
person I know of in town.  He and I work together and we have multiple
existing Asterisk setups in our office along with analog and digital
phones.   That makes for an easier "get things going" setup for the
first few meetings I think.  At any meeting if we find our facilities
too small or lacking, we can move the next meeting to the other
location.

Wayne

On Wed, Oct 12, 2005 at 07:30:33AM -0500, Chris Tooley wrote:
> Having heard no complaints about the idea of at least doing the first
> meeting here, can I assume that's acceptable?
> 
> -- 
> Chris Tooley
> 512-646-1507
> ctooley@gnumber.com
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 

Wayne Walker

wwalker@gnumber.com                   Do you use Linux?!
http://www.gnumber.com                Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
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On Wed, 2005-10-12 at 07:53 -0500, Les Elliott wrote:
> Where is the office

3925 W Braker Ln (The MCC Building at MoPac and Braker)

> and when do you plan to meet?

That was the next order of business, when can everyone get away to have
our first gathering.

> Les
> 
> On 10/12/05, Chris Tooley <chris@tooley.com> wrote:
>         Having heard no complaints about the idea of at least doing
>         the first
>         meeting here, can I assume that's acceptable?
>         
>         --
>         Chris Tooley
>         512-646-1507
>         ctooley@gnumber.com 
>         
>         _______________________________________________
>         Austin-Asterisk-Users-Group mailing list
>         Austin-Asterisk-Users-Group@bybent.com
>         http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 

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Works for me.  What time?

On Wed, 2005-10-12 at 08:30 -0500, Chris Tooley wrote:
> > That was the next order of business, when can everyone get away to have
> > our first gathering.
> 
> How is Monday the 17th for everyone?  Third Monday of the month
> schedule?
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

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From: David Bluestein II <dbii@interaction.net>
Subject: Re: [A*UG] Welcome, First Meeting
Date: Wed, 12 Oct 2005 08:58:22 -0500
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I think that sounds good. Unless someone else has a comparable setup 
(which I don't know of anyone).

Maybe we could even get the O'Reilly Asterisk book writer to come 
sometime.

David

On Oct 12, 2005, at 8:47 AM, Wayne Walker wrote:

> First real posting!
>
> While another member offered his office space (a little larger than
> ours) for meetings, Chris Tooley is by far the most knowlegable 
> Asterisk
> person I know of in town.  He and I work together and we have multiple
> existing Asterisk setups in our office along with analog and digital
> phones.   That makes for an easier "get things going" setup for the
> first few meetings I think.  At any meeting if we find our facilities
> too small or lacking, we can move the next meeting to the other
> location.
>
> Wayne
>
> On Wed, Oct 12, 2005 at 07:30:33AM -0500, Chris Tooley wrote:
>> Having heard no complaints about the idea of at least doing the first
>> meeting here, can I assume that's acceptable?
>>
>> -- 
>> Chris Tooley
>> 512-646-1507
>> ctooley@gnumber.com
>>
>> _______________________________________________
>> Austin-Asterisk-Users-Group mailing list
>> Austin-Asterisk-Users-Group@bybent.com
>> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
>
> -- 
>
> Wayne Walker
>
> wwalker@gnumber.com                   Do you use Linux?!
> http://www.gnumber.com                Get Counted!  
> http://counter.li.org/
> Perl - http://www.perl.org/           Perl User Groups - 
> http://www.pm.org/
> Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
> IRC:     wwalker on freenode.net
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
>
-----------------------------------------------------------------------
David H. Bluestein II
President & Lead Developer         dbii@interaction.net
ii, inc.                                                  
http://www.interaction.net
     -- Specializing in Interactive, Database Driven Websites --

From john_g@cibolo.com Wed Oct 12 11:53:55 2005
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That sounds good to me.

Thanks for offering the knowhow and the list
initiative!

How much of a server can asterisk run on?
For a minimal small bus. 3 phone
setup, could one of Dzuy's costlinux servers
based on either old netier boxes, or new generic fanless
mini-ITX motherboards in generic small boxes do the
serving to some IP phones with a processor clock speed of 200-300MHz?

John Griessen

On Wed, 2005-10-12 at 08:30 -0500, Chris Tooley wrote:
> > That was the next order of business, when can everyone get away to have
> > our first gathering.
>
> How is Monday the 17th for everyone?  Third Monday of the month
> schedule?
>
From wwalker@behemoth.gnumber.com Wed Oct 12 12:04:26 2005
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From: Wayne Walker <wwalker@bybent.com>
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Is everyone OK with Austin Asterisk Users Group?  I don't live in
Austin, but it's shorter than Central Texas or Capitol Area or whatever.  

We can change it later.  Say at Monday's meeting...

-- 

Wayne Walker

wwalker@bybent.com                    Do you use Linux?!
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On Wed, Oct 12, 2005 at 11:46:42AM -0500, John Griessen wrote:
> That sounds good to me.
> 
> Thanks for offering the knowhow and the list
> initiative!
> 
> How much of a server can asterisk run on?
A LinkSys WRT54G wireless router (70 USD) can run asterisk, and handles
about 5 simultaneous calls!  Any reasonable number of extensions, but
only 4 to 7 simulataneous calls I think.

So the slowest desktop computer you have will handle more than you need.

> For a minimal small bus. 3 phone
> setup, could one of Dzuy's costlinux servers
> based on either old netier boxes, or new generic fanless
> mini-ITX motherboards in generic small boxes do the
> serving to some IP phones with a processor clock speed of 200-300MHz?
Yes.

> 
> John Griessen
> 
> On Wed, 2005-10-12 at 08:30 -0500, Chris Tooley wrote:
> > > That was the next order of business, when can everyone get away to have
> > > our first gathering.
> >
> > How is Monday the 17th for everyone?  Third Monday of the month
> > schedule?
> >
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 

Wayne Walker

wwalker@gnumber.com                   Do you use Linux?!
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From: Wayne Walker <wwalker@bybent.com>
To: Austin Asterisk Users Group <austin-asterisk-users-group@bybent.com>
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I was looking at the membership list and I think George and Taylor may
not be receiving any messages to the list.  

Taylor, George, please reply to me if you get this.


-- 

Wayne Walker

wwalker@bybent.com                    Do you use Linux?!
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From chris@tooley.com Wed Oct 12 14:37:58 2005
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So, the plan is currently to have the meeting at Wayne Walker and Chris
Tooley's office at the MCC Building.  The address is 3925 W Braker Lane
in Austin.  When you arrive the side of the building that faces Braker
is the back of the building (which is the correct side).  When
approaching the door at the back of the building there will be a grey
box containing a phone.  If you pick up the phone and dial 96461507
you'll be connected to my office and I will come get you.  The building
is "secured" so someone will have to come get everyone (which shouldn't
be that bad).

For timing we're thinking 6:30PM.

-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From gmahan@dmibs.com Wed Oct 12 14:57:29 2005
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Date: Wed, 12 Oct 2005 14:50:39 -0500
Subject: Re: [A*UG] checking
From: George Mahan <gmahan@dmibs.com>
To: Wayne Walker <wwalker@bybent.com>,
   Austin Asterisk Users Group <austin-asterisk-users-group@bybent.com>
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I received it...
-- 
George Mahan, CCIE #4677
Digital Minds Internet Business Solutions
Office: 512.276.5138
Cell: 512.924.1243
gmahan@dmibs.com
www.dmibs.com





> From: Wayne Walker <wwalker@bybent.com>
> Reply-To: Wayne Walker <wwalker@bybent.com>
> Date: Wed, 12 Oct 2005 14:17:28 -0500
> To: Austin Asterisk Users Group <austin-asterisk-users-group@bybent.com>
> Subject: [A*UG] checking
> 
> I was looking at the membership list and I think George and Taylor may
> not be receiving any messages to the list.
> 
> Taylor, George, please reply to me if you get this.
> 
> 
> -- 
> 
> Wayne Walker
> 
> wwalker@bybent.com                    Do you use Linux?!
> http://www.bybent.com                 Get Counted!  http://counter.li.org/
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From john_g@cibolo.com Wed Oct 12 23:56:25 2005
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Is there more than one driveway?  Is the drive to the back
side marked some way?

I remember ending up at the "front side"
by default pretty easily when I have been there...

John G

Chris Tooley wrote:
When you arrive the side of the building that faces Braker
> is the back of the building (which is the correct side).  When
> approaching the door at the back of the building there will be a grey
> box containing a phone.  
From fmouse@fmp.com Thu Oct 13 00:09:32 2005
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When and where?  I missed the first part of this thread.

On Wed, 2005-10-12 at 23:52 -0500, John Griessen wrote:
> Is there more than one driveway?  Is the drive to the back
> side marked some way?
> 
> I remember ending up at the "front side"
> by default pretty easily when I have been there...
> 
> John G
> 
> Chris Tooley wrote:
> When you arrive the side of the building that faces Braker
> > is the back of the building (which is the correct side).  When
> > approaching the door at the back of the building there will be a grey
> > box containing a phone.  
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | http://pubkeys.fmp.com
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |


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What's the date and time on this?  I missed the 1st part of the thread.

Thus spake John Griessen on Wed, Oct 12, 2005 at 11:52:19PM CDT
> Is there more than one driveway?  Is the drive to the back
> side marked some way?
> 
> I remember ending up at the "front side"
> by default pretty easily when I have been there...
> 
> John G
> 
> Chris Tooley wrote:
> When you arrive the side of the building that faces Braker
> > is the back of the building (which is the correct side).  When
> > approaching the door at the back of the building there will be a grey
> > box containing a phone.  
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
From ctooley@gnumber.com Thu Oct 13 09:14:50 2005
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If you take the driveway from Braker Lane (the driveway is on the south
side of Braker closest to MoPac) and then take the very first left (it
looks like you're turning into the loading docks), you will be at the
back entrance.

Lindsay:

Monday October 17th at 6:30 PM

On Wed, 2005-10-12 at 23:52 -0500, John Griessen wrote:
> Is there more than one driveway?  Is the drive to the back
> side marked some way?
> 
> I remember ending up at the "front side"
> by default pretty easily when I have been there...
> 
> John G
> 
> Chris Tooley wrote:
> When you arrive the side of the building that faces Braker
> > is the back of the building (which is the correct side).  When
> > approaching the door at the back of the building there will be a grey
> > box containing a phone.  
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From wwalker@behemoth.gnumber.com Thu Oct 13 09:17:57 2005
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From: Wayne Walker <wwalker@bybent.com>
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Subject: Re: [A*UG] Welcome, First Meeting
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Monday, October 17th,
at 6:30 PM
at the gNumber office
in the MCC building
at the southwest corner of MoPac and Braker.

The "back" is the North side.  If you can't look across the street and
see Burger King, you are not on the back/north side.

And the parking lot is a twisty little maze.  Good luck :)  But it's a
mostly open maze and Burger King is your signpost.

Wayne


On Wed, Oct 12, 2005 at 11:52:19PM -0500, John Griessen wrote:
> Is there more than one driveway?  Is the drive to the back
> side marked some way?
> 
> I remember ending up at the "front side"
> by default pretty easily when I have been there...
> 
> John G
> 
> Chris Tooley wrote:
> When you arrive the side of the building that faces Braker
> > is the back of the building (which is the correct side).  When
> > approaching the door at the back of the building there will be a grey
> > box containing a phone.  
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 

Wayne Walker

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
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From ctooley@gnumber.com Thu Oct 13 09:22:41 2005
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Subject: Re: [A*UG] Welcome, First Meeting
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For 2 people who work across the hall from each other, we duplicate a
lot of each other's work. :)  You'd think we'd be better at coordinating
things.

On Thu, 2005-10-13 at 09:17 -0500, Wayne Walker wrote:
> Monday, October 17th,
> at 6:30 PM
> at the gNumber office
> in the MCC building
> at the southwest corner of MoPac and Braker.
> 
> The "back" is the North side.  If you can't look across the street and
> see Burger King, you are not on the back/north side.
> 
> And the parking lot is a twisty little maze.  Good luck :)  But it's a
> mostly open maze and Burger King is your signpost.
> 
> Wayne
> 
> 
> On Wed, Oct 12, 2005 at 11:52:19PM -0500, John Griessen wrote:
> > Is there more than one driveway?  Is the drive to the back
> > side marked some way?
> > 
> > I remember ending up at the "front side"
> > by default pretty easily when I have been there...
> > 
> > John G
> > 
> > Chris Tooley wrote:
> > When you arrive the side of the building that faces Braker
> > > is the back of the building (which is the correct side).  When
> > > approaching the door at the back of the building there will be a grey
> > > box containing a phone.  
> > _______________________________________________
> > Austin-Asterisk-Users-Group mailing list
> > Austin-Asterisk-Users-Group@bybent.com
> > http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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Maybe your binary:)

On Thu, 2005-10-13 at 09:22 -0500, Chris Tooley wrote:
> For 2 people who work across the hall from each other, we duplicate a
> lot of each other's work. :)  You'd think we'd be better at coordinating
> things.
> 
> On Thu, 2005-10-13 at 09:17 -0500, Wayne Walker wrote:
> > Monday, October 17th,
> > at 6:30 PM
> > at the gNumber office
> > in the MCC building
> > at the southwest corner of MoPac and Braker.
> > 
> > The "back" is the North side.  If you can't look across the street and
> > see Burger King, you are not on the back/north side.
> > 
> > And the parking lot is a twisty little maze.  Good luck :)  But it's a
> > mostly open maze and Burger King is your signpost.
> > 
> > Wayne
> > 
> > 
> > On Wed, Oct 12, 2005 at 11:52:19PM -0500, John Griessen wrote:
> > > Is there more than one driveway?  Is the drive to the back
> > > side marked some way?
> > > 
> > > I remember ending up at the "front side"
> > > by default pretty easily when I have been there...
> > > 
> > > John G
> > > 
> > > Chris Tooley wrote:
> > > When you arrive the side of the building that faces Braker
> > > > is the back of the building (which is the correct side).  When
> > > > approaching the door at the back of the building there will be a grey
> > > > box containing a phone.  
> > > _______________________________________________
> > > Austin-Asterisk-Users-Group mailing list
> > > Austin-Asterisk-Users-Group@bybent.com
> > > http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> > 

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All this Asterisk talk has piqued my interest!

At my home/business, I have a needed for two or more intercom stations. 
The station sites are ~2000' apart, but are on my LAN.  I used to use 
some unusual telephones that gave intercom service when two telephone 
lines were attached.  At one time, I had 6 or 8 of those telephones; 
they have all died and seem to be not replaceable.  Especially if I can 
get voip working on my satellite (net2dish/dirceway) broadband, I would 
like to drop one of my phone lines.  Or both.  Alas, my experiments 
with Skype indicate that the latency is insufferable.

So, here are my questions:
1) Can Asterisk give me intercom service over my LAN.  I'm almost sure 
the answer to that is "yes"; if so, what hardware do I need?
2) I assume I can use regular ole telephone handsets for the intercom?
3) Once I have the intercom service setup, is it trivial to then use 
the intercom stations to access my telephone line?

It seems attractive to find preconfigured and tested boxes to provide 
the above capabilities.  Comments?  Recommendations?

Muchas gracias!

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  472 days 22 hours 50 minutes
From wwalker@behemoth.gnumber.com Thu Oct 13 13:17:44 2005
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Date: Thu, 13 Oct 2005 12:18:09 -0500
From: Wayne Walker <wwalker@bybent.com>
To: Chris Tooley <ctooley@gnumber.com>
Subject: Re: [A*UG] Welcome, First Meeting
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On Thu, Oct 13, 2005 at 09:14:49AM -0500, Chris Tooley wrote:
> If you take the driveway from Braker Lane (the driveway is on the south
> side of Braker closest to MoPac) and then take the very first left (it
> looks like you're turning into the loading docks), you will be at the
> back entrance.
When you take the left he mentions above, you need to take the next left
also (about 20 feet), that's a curving road and you can turn right into
either of the two back parking lots.

Wayne

-- 

Wayne Walker

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net
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It was a pleasure meeting all of you at the first meeting of the A*UG 
last night.  Newbies like me got a great overview and a lot of useful 
information.  My wife thanked you for providing answers to her SNOT list 
questions.

I have downloaded Asterisk@Home and will be playing with it in a day or 
so to start getting practical experience.  First I'll dip my toes into a 
simple PBX with just a few POTS stations in the house, then mix in a 
little VOIP.

Also ordered the O'Reilly book.

After a while perhaps I'll be ready to start thinking about an overall 
design. This will be much easier with local contacts to ease the way 
over the expected hard spots.

It's also nice being a charter member of something.

Larry 301-1414

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From lelliott5@gmail.com Tue Oct 18 18:15:34 2005
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Subject: Re: [A*UG] A most excellent first meeting
From: Les Elliott <lelliott5@gmail.com>
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Right on!
Les

On Tue, 2005-10-18 at 16:36 -0500, Larry Alkoff wrote:
> It was a pleasure meeting all of you at the first meeting of the A*UG 
> last night.  Newbies like me got a great overview and a lot of useful 
> information.  My wife thanked you for providing answers to her SNOT list 
> questions.
> 
> I have downloaded Asterisk@Home and will be playing with it in a day or 
> so to start getting practical experience.  First I'll dip my toes into a 
> simple PBX with just a few POTS stations in the house, then mix in a 
> little VOIP.
> 
> Also ordered the O'Reilly book.
> 
> After a while perhaps I'll be ready to start thinking about an overall 
> design. This will be much easier with local contacts to ease the way 
> over the expected hard spots.
> 
> It's also nice being a charter member of something.
> 
> Larry 301-1414
> 

From fc4linux@kojetin.org Tue Oct 18 20:10:28 2005
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Hello everybody,

 I realize there was a meeting that was held recently.

How appropriate is it for me to attend these meetings if I don't really 
know anyone personally on the CTLUG or A*UG? Also I am not a Linux guru 
by any measure, but I love to learn and have made a lot of progress!

I have the Asterisk PBX beta running on my FC4 box and am actually 
learning it quite quickly. After all there is only about 30 config 
files.  It is working though and I plan to purchase an IP phone sometime 
in the future to use with it. Anyone have any suggestion for a good geek 
ip phone?

Maybe I will meet some of you guru's in the future.

Michael Kojetin

From svaughan@austin.rr.com Tue Oct 18 20:12:55 2005
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  Sounds awesome.  I work just down the road (Kramer/Metric) but was 
unable to
make it to that particular meeting.  Can anyone shoot a summary of what 
went down?

TIA,
-steve

Larry Alkoff wrote:

> It was a pleasure meeting all of you at the first meeting of the A*UG 
> last night.  Newbies like me got a great overview and a lot of useful 
> information.  My wife thanked you for providing answers to her SNOT 
> list questions.
>
> I have downloaded Asterisk@Home and will be playing with it in a day 
> or so to start getting practical experience.  First I'll dip my toes 
> into a simple PBX with just a few POTS stations in the house, then mix 
> in a little VOIP.
>
> Also ordered the O'Reilly book.
>
> After a while perhaps I'll be ready to start thinking about an overall 
> design. This will be much easier with local contacts to ease the way 
> over the expected hard spots.
>
> It's also nice being a charter member of something.
>
> Larry 301-1414
>

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From: Wayne Walker <wwalker@bybent.com>
To: Michael Kojetin <fc4linux@kojetin.org>
Subject: Re: [A*UG] I didn't attend... can I?
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On Tue, Oct 18, 2005 at 08:05:21PM -0500, Michael Kojetin wrote:
> Hello everybody,
> 
> I realize there was a meeting that was held recently.
> 
> How appropriate is it for me to attend these meetings if I don't really 
> know anyone personally on the CTLUG or A*UG? Also I am not a Linux guru 
> by any measure, but I love to learn and have made a lot of progress!
None of us knew each other when we first showed up.  Come on down!!

> I have the Asterisk PBX beta running on my FC4 box and am actually 
> learning it quite quickly. After all there is only about 30 config 
> files.  It is working though and I plan to purchase an IP phone sometime 
> in the future to use with it. Anyone have any suggestion for a good geek 
> ip phone?
> 
> Maybe I will meet some of you guru's in the future.
Hope to see you soon!  Though I'm no guru.
> 
> Michael Kojetin
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 

Wayne Walker

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From: Wayne Walker <wwalker@bybent.com>
To: Willie McKemie <mckemie@spamcop.net>
Subject: Re: [A*UG] Advice
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On Thu, Oct 13, 2005 at 10:59:39AM -0500, Willie McKemie wrote:
> All this Asterisk talk has piqued my interest!
> 
> At my home/business, I have a needed for two or more intercom stations. 
> The station sites are ~2000' apart, but are on my LAN.  I used to use 
> some unusual telephones that gave intercom service when two telephone 
> lines were attached.  At one time, I had 6 or 8 of those telephones; 
> they have all died and seem to be not replaceable.  Especially if I can 
> get voip working on my satellite (net2dish/dirceway) broadband, I would 
> like to drop one of my phone lines.  Or both.  Alas, my experiments 
> with Skype indicate that the latency is insufferable.
> 
> So, here are my questions:
> 1) Can Asterisk give me intercom service over my LAN.  I'm almost sure 
> the answer to that is "yes"; if so, what hardware do I need?
Intercom?  not exactly.  Phones, yes. You would need either a VoIP phone
($60 to $200 each) or an ATA (ethernet plug on one side and an RJ-11
phone jack on the other side) ($80-$200 each) at each location.

If you have two buildings and just need service between them, you could
have one ATA at each building and put as many phones as you like.

Too bad you missed the meeting, there are so many forks in the
conversation based on your answers, it's hard to handle via email.

> 2) I assume I can use regular ole telephone handsets for the intercom?
With an ATA yes.  With an asterisk server with an FXS card in it, yes.

> 3) Once I have the intercom service setup, is it trivial to then use 
> the intercom stations to access my telephone line?
Yes, but...

What you would really do is just what a normal business does.  You have
one or more lines coming in from your phone provider (whether it's SBC,
GTE, or a VoIP provider).  These connect into your Asterisk server.
Then your asterisk server acts as a PBX.  You then have "extensions"
each extensions is either an ATA (Analog Telephone Adapter), an FXS port,
or a VoIP phone.  An ATA officially drives one phone line, but usually
works fine with say up to 5.  Same with an FXS port.  ATA's and FXS
ports run ~$80-150 each.

> It seems attractive to find preconfigured and tested boxes to provide 
> the above capabilities.  Comments?  Recommendations?
I've heard some good about "asterisk@home" which is an installer that
will WIPE your hard drive, install linux, and asterisk, and configure
asterisk somewhat.

I'd recommend running RHEL4, CentOS4, or FC4 on a machine and installing
Asterisk manually.

Make the next meeting and we can sort things out with you.

-- 

Wayne Walker

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
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Subject: Re: [A*UG] Advice
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Thus spake Wayne Walker on Tue, Oct 18, 2005 at 09:58:24PM CDT
> On Thu, Oct 13, 2005 at 10:59:39AM -0500, Willie McKemie wrote:
> > All this Asterisk talk has piqued my interest!
> > 
> > At my home/business, I have a needed for two or more intercom stations. 
> > The station sites are ~2000' apart, but are on my LAN.  I used to use 
> > some unusual telephones that gave intercom service when two telephone 
> > lines were attached.  At one time, I had 6 or 8 of those telephones; 
> > they have all died and seem to be not replaceable.  Especially if I can 
> > get voip working on my satellite (net2dish/dirceway) broadband, I would 
> > like to drop one of my phone lines.  Or both.  Alas, my experiments 
> > with Skype indicate that the latency is insufferable.
> > 
> > So, here are my questions:
> > 1) Can Asterisk give me intercom service over my LAN.  I'm almost sure 
> > the answer to that is "yes"; if so, what hardware do I need?
> Intercom?  not exactly.  Phones, yes. You would need either a VoIP phone
> ($60 to $200 each) or an ATA (ethernet plug on one side and an RJ-11
> phone jack on the other side) ($80-$200 each) at each location.

When you think of Asterisk, think of PBX - like a traditional POTS
telephone PBX, but completely programmable.  Depending on how much you want
to invest in hardware, you can feed as many POTS lines to the world (FXS) or
phones (FXO) as you can afford the hardware to support.  You can also access
VoIP services on the Internet from Asterisk-friendly VoIP providers, kind of
like a "digital FXS" line - sort of.  I don't expect that Asterisk would be
necessary for digital intercom over a LAN.  This sounds more like the domain
of some other VoIP software, kind of like audio IM.

"To listen to a duck, press 9 now"

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
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From: Willie McKemie <mckemie@spamcop.net>
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Thanks for responding!

On Tue, Oct 18, 2005 at 09:58:24PM -0500, Wayne Walker wrote:
> On Thu, Oct 13, 2005 at 10:59:39AM -0500, Willie McKemie wrote:
> > All this Asterisk talk has piqued my interest!
> 
> > 3) Once I have the intercom service setup, is it trivial to then use 
> > the intercom stations to access my telephone line?
> Yes, but...
> 
> What you would really do is just what a normal business does.  You have
> one or more lines coming in from your phone provider (whether it's SBC,
> GTE, or a VoIP provider).  These connect into your Asterisk server.
> Then your asterisk server acts as a PBX.  You then have "extensions"
> each extensions is either an ATA (Analog Telephone Adapter), an FXS port,
> or a VoIP phone.  An ATA officially drives one phone line, but usually
> works fine with say up to 5.  Same with an FXS port.  ATA's and FXS
> ports run ~$80-150 each.

I'm now developing a little bit better mental model of how this stuff 
works.  I've ordered a couple of IP telephones and a couple of clone 
cards for Asterisk.  Tuesday, I looked at Frys for IP telephones, I 
found some ATAs, but no telephones.
> 
> > It seems attractive to find preconfigured and tested boxes to provide 
> > the above capabilities.  Comments?  Recommendations?
> I've heard some good about "asterisk@home" which is an installer that
> will WIPE your hard drive, install linux, and asterisk, and configure
> asterisk somewhat.
> 
> I'd recommend running RHEL4, CentOS4, or FC4 on a machine and installing
> Asterisk manually.

I have a bunch of Debian boxes and I notice that asterisk is in the 
Debian repositories; I'll start there since I have no RH boxes.  If 
that doesn't go well, I might try asterisk@home on a disposable box.
> 
> Make the next meeting and we can sort things out with you.

I live a ways out and very rarely go to town in the evenings. 

BTW, I subscribed to the digest list as either 
"sprinter@austinfarm.org" or maybe "mckemie@austinfarm.org".  I 
received one digest about a week ago, but none since.  Don't know 
whether the list is inactive or my subscription is screwed up.

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  478 days 17 hours 21 minutes
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Sorry I missed the first meeting.  I wanted to come but I live kinda far out
of town.  I may be able to make the next one.

Suggestion:  Lenny Tropino and Brian Sinclair have given a couple of really
dynamite hands-on presentations on VoIP at CACTUS meetings.  They run
VoIPing (http://www.voiping.com) and have lots of experience with this
stuff.  I'm sure they'd be more than willing to come and talk to the A*UG
folken and do the same thing.  I know them and if you wish, I'll make the
contact and inquire.

You can run through the formal parts of their presentation at
<http://www.voiping.com/cactus/>.

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
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Lenny, Brian,

If you don't know already, a bunch of folks have formed the Austin Asterisk
User's Group.  A lot of these people are also in the Central Texas Linux
Users Group, whose website and mailing list I host.  They had their very
first meeting last Monday.  I wasn't there, but I'm on the A*UG mailing
list, and suggested that perhaps A*UG could get y'all to come and do
something similar to what you did for CACTUS some time ago.  Don't know how
many people are involved, but they're very enthusiastic.  I ain't in charge
of programs for the group, and this is only a suggestion to them, but I
thought I'd see if I could facilitate this if you were willing and A*UG were
interested.

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
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Subject: Re: [A*UG] VoiPIng
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Lindsay Haisley wrote:

>Sorry I missed the first meeting.  I wanted to come but I live kinda far out
>of town.  I may be able to make the next one.
>
>Suggestion:  Lenny Tropino and Brian Sinclair have given a couple of really
>dynamite hands-on presentations on VoIP at CACTUS meetings.  They run
>VoIPing (http://www.voiping.com) and have lots of experience with this
>stuff.  I'm sure they'd be more than willing to come and talk to the A*UG
>folken and do the same thing.  I know them and if you wish, I'll make the
>contact and inquire.
>
>You can run through the formal parts of their presentation at
><http://www.voiping.com/cactus/>.
>
>  
>
I would be most enthusiastic to hear their presentation. I hope it works 
out.

Michael
From lenny@VoIPing.com Wed Oct 19 20:11:03 2005
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Subject: [A*UG] Hello from VoIPing...
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Folks,

Hello!  Found out Austin has a newly formed Asterisk User Group.  Let
me introduce myself (Lindsey already did, sorta).  I've personally been
using/playing with Asterisk for about 3+ years (yea, early days of
the code).  When VoIPing was fully engaged (we're only taking select
clients now) we had several VoIP customers in town.  We did a larger
install/customization at a local company (60+ phones) and when I worked
at another startup, I replaced their TDM PBX (that was undersized) with
a complete Asterisk PBX.  

Although I don't have a lot of recent time to be playing with Asterisk,
I keep current on the comings and goings of the project.  A good friend
of mine, Gil Kloepfer, is also extremely versed with Asterisk and
has done several presentations at CACTUS.  Gil got interested in Asterisk
through my interactions and has recently taken it to the "n-th degree" at
his current employer.  Gil is an active bug-tracker/submitter and has
made FreeBSD compatibility even more successful.

Last year, at the first annual Astricon, I was a speaker and it had
a very large turnout.  The 2nd annual (USA) Astricon just recently 
happened, but I was unable to attend.

Presentations from that event are available at:
	http://www.astricon.net/conf2004/program.shtml
Specifically mine was:
	http://www.astricon.net/conf2004/docs/3-lenny-tropiano.pdf
Photos from that event are available at: 
	http://photos.tropiano.org/gallery/astricon-2004

Gil, Brian and myself have Asterisk Installations at home using
Cisco, Polycom, SoftPhones, Sipuras, and IAX/SIP providers.

I'd be happy to do a presentation at an upcoming A*UG ... 

Thanks for starting this group, I'll be e-mailing some of my cohorts
and letting them know about it, as well.

Talk to everyone soon,
Lenny

-- 
Lenny Tropiano                          E-mail: lenny@voiping.com
Partner, Networking Specialist          Pager:  pager-lenny@voiping.com
VoIPing, LLC                            URL:    http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-0867   Mobile: 512-698-VOIP [8647] 
From ctooley@gnumber.com Thu Oct 20 07:32:47 2005
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> I'd be happy to do a presentation at an upcoming A*UG ... 
> 
Definitely come to a meeting, and join in the discussion.  We haven't
really set a formal format for the meetings but discussion, assistance
and information dissemination is a more likely format for a while than
formal presentations.  We're mostly non-telephony people just getting
into Asterisk, with a couple of people who make a living doing it.

> Thanks for starting this group, I'll be e-mailing some of my cohorts
> and letting them know about it, as well.
> 
We'd be happy to see them.

> Talk to everyone soon,
> Lenny
> 
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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We're currently doing intra-office paging via Meetme conferences and the
auto-answer feature available in Polycom phones.  You dial a paging
extension, say something into it, hang up, and it broadcasts the message to
all the phones.  Works great so far.

We have a prospect who also needs overhead paging via their existing paging
speakers.  Does anybody have any experience with such systems or
recommendations on how to integrate Asterisk with overhead speakers?

Thanks,
Carey

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If the current overhead speakers are used via a telephony interface and
FXO or FXS channel should give you what you need (it's unlikely that
it's a digital channel).  If it's an audio input, like speaker jacks,
you can use a sound card and OSS/ALSA as a channel as well.

Once you've got the channel, you can use it just like you do the Meetme.
But more likely you'll want to dial the number, have that
context/extension open an OSS/ALSA channel and have them start talking.
Otherwise if you need record and them playback I can show you some
examples of how I did that for recording prompts that seems to work.

On Thu, 2005-10-20 at 08:21 -0500, Carey Jung wrote:
> We're currently doing intra-office paging via Meetme conferences and the
> auto-answer feature available in Polycom phones.  You dial a paging
> extension, say something into it, hang up, and it broadcasts the message to
> all the phones.  Works great so far.
> 
> We have a prospect who also needs overhead paging via their existing paging
> speakers.  Does anybody have any experience with such systems or
> recommendations on how to integrate Asterisk with overhead speakers?
> 
> Thanks,
> Carey
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From pehr@alumni.utexas.net Thu Oct 20 19:04:48 2005
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Subject: [A*UG] Welcome, First Meeting - DOH!
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I've been struggling with Asterisk since sometime last week.  Trying  
to use it as a PBX for our new office.  I just can't believe my  
luck.  Here it is Thursday and the first meeting of A*UG was three  
days ago. Doh!  I'll be there next month unless I've succumbed to the  
temptation of calling some vendor of an overpriced commercial solution.

If anyone is interested: we are running Asterisk on an AMD based  
eMachines box - AMD Athlon 2400+.   I have a Digium TDM02B (TDM400P +  
2 PORT FXO).  We have two Grandstream BudgeTone telephones.  It is  
all sort of working.  But receivers of calls over the PSTN complain  
that we sound muffled and that the volume is a bit low.  Inside the  
office, i.e., calls that never leaves our intranet, sound OK.  Thus,  
I am suspecting that it is the FXO interface that is not tweaked  
right.  We previously had some cheaper cards that we bought from  
Digitnetworks.  For those I tweaked RXGAIN and TXGAIN in zapata.conf.  
That did not seem to work.  I have now reset zapata.conf to the  
original values:

;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300        ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

;Include AMP configs
#include zapata_additional.conf

If anyone has suggestions on what to try, I am all ears.

Pehr

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Digium has great support for their cards.  If you call them, they will
resolve these issues for even ssh'ing to the machine and connecting to a
shared screen session so that you can see what they're doing.

On Thu, 2005-10-20 at 18:58 -0500, Pehr Jansson wrote:
> 
> I've been struggling with Asterisk since sometime last week.  Trying  
> to use it as a PBX for our new office.  I just can't believe my  
> luck.  Here it is Thursday and the first meeting of A*UG was three  
> days ago. Doh!  I'll be there next month unless I've succumbed to
> the  
> temptation of calling some vendor of an overpriced commercial
> solution.
> 
> If anyone is interested: we are running Asterisk on an AMD based  
> eMachines box - AMD Athlon
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From fc4linux@kojetin.org Thu Oct 20 23:56:15 2005
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From: Michael Kojetin <fc4linux@kojetin.org>
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Subject: [A*UG] My * status
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I figured I would update all as to how my progress with * is coming 
along at home.

Last week I downloaded the 1.2 beta, compiled and installed it with no 
problems. I also read as much documentation and case studies as possible 
which really helped me in understanding how the program works. After 
about 3 or so hours of fiddling around with I went from completely 
ignorant about Asterisk to making successful SIP and IAX calls from 
computer to computer 1x1pbx.

Like David McNett said on his website, there is only so much you can do 
with the desktop to laptop calling so I signed up for Voicepulse and now 
got that working. My first call had horrible quality so I took David's 
advice and switched the outgoing Voicepulse servers around so it uses 
the backup server as the primary. After I switched the servers around, I 
called my family in Montana and was experiencing perfect call quality 
for the two hours we talked using the IAX Firefly soft phone. 

Next up on the list of things to do is purchase an actual IP phone. Not 
exactly sure what I want to go with. I don't really have much use for 
the Cisco 7960 though it is extremely cool and does have a lot of 
features. I am leaning toward something like a wifi ip phone. Anyone 
have experience with these?  On David's website he had a horrible 
experience with the ZyXEL model. I surly don't want to repeat that but 
have heard the new firmware does a lot to fix the previous problems. 
I'll search around for a different model.

Now all I have to do is to make my dialplan be useful....


From fc4linux@kojetin.org Fri Oct 21 00:01:52 2005
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Subject: [A*UG] Asterisk: The Future of Telephony, O'Reilly 
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Hello all,

Thought I would share this bit of info. I'm not sure if anyone else 
knows about it or not but I guess the O'Reilly Asterisk book is freely 
available online. I am reading it now and its a good intro...

 http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

I guess O'Reilly Media agreed to it...


Michael


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Subject: [A*UG] How to wire Asterisk to stations?
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I've just finished the O-Reilly book on Asterisk and setup a separate 
computer with Asterisk@Home.

The book has a lot of good information but for some reason does not 
discuss the physical wiring that goes to stations (FXS) and FXOs.

I understand that cat5 type wire would be used.
What commonly used standards, if any, are used on the various colored 
wires?  What signals and voltages to they carry?

Presently I have a lot of POTS telephones all wired in 'home run' with a 
single cat5 from line 1 of a station RJ-11 station block to type 66 
blocks where the Asterisk equipment will live.

Can I do the same with Asterisk station lines - that is each extension 
would be a single pair or two from Asterisk that carries power, ringing 
current and voice.

What equipment do I need at the Asterisk end to connect multiple analog 
POTS phone each station?

I also need to know how many wire pairs are needed for non-POTS phones 
and what those wire carry.

If feasable the easiest wiring method is to end up with all POTS phones 
on the default inner pair line 1 (blue) and non-POTS phones on line 2 
(orange) and if necessary, 3 (green) leaving 4 (extra pair possibly used 
for direct connection to CO line for emergencies).  That way any single 
(not double) line POTS phone can be safely plugged in as they are all on 
line 1.  Could this be done?

My mnemonic for telephone wire color coding is "BOG Brown".

I'm interested in any comments on physical wiring.

Larry
-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From fmouse-aaug@fmp.com Fri Oct 21 15:44:58 2005
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From: Lindsay Haisley <fmouse-aaug@fmp.com>
To: austin-asterisk-users-group@bybent.com
Subject: Re: [A*UG] How to wire Asterisk to stations?
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I believe the IP phones use regular cat5 ethernet lines, yes?  Treat them on
the network as if they were computers connected through a hub.  You'll need
to address the appropriate traffic to the interface on your Asterisk box.

POTS phones, of course, use single twisted pairs, and if you use cat5 you
can get 4 phone lines in a single cat5 cable and wire them any way you wish. 
There's no magic there.

Thus spake Larry Alkoff on Fri, Oct 21, 2005 at 03:39:22PM CDT
> I've just finished the O-Reilly book on Asterisk and setup a separate 
> computer with Asterisk@Home.
> 
> The book has a lot of good information but for some reason does not 
> discuss the physical wiring that goes to stations (FXS) and FXOs.
> 
> I understand that cat5 type wire would be used.
> What commonly used standards, if any, are used on the various colored 
> wires?  What signals and voltages to they carry?
> 
> Presently I have a lot of POTS telephones all wired in 'home run' with a 
> single cat5 from line 1 of a station RJ-11 station block to type 66 
> blocks where the Asterisk equipment will live.
> 
> Can I do the same with Asterisk station lines - that is each extension 
> would be a single pair or two from Asterisk that carries power, ringing 
> current and voice.
> 
> What equipment do I need at the Asterisk end to connect multiple analog 
> POTS phone each station?
> 
> I also need to know how many wire pairs are needed for non-POTS phones 
> and what those wire carry.
> 
> If feasable the easiest wiring method is to end up with all POTS phones 
> on the default inner pair line 1 (blue) and non-POTS phones on line 2 
> (orange) and if necessary, 3 (green) leaving 4 (extra pair possibly used 
> for direct connection to CO line for emergencies).  That way any single 
> (not double) line POTS phone can be safely plugged in as they are all on 
> line 1.  Could this be done?
> 
> My mnemonic for telephone wire color coding is "BOG Brown".
> 
> I'm interested in any comments on physical wiring.
> 
> Larry
> -- 
> Larry Alkoff N2LA - Austin TX
> Using Thunderbird on Slackware Linux
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
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Subject: Re: [A*UG] How to wire Asterisk to stations?
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Thus spake Lindsay Haisley on Fri, Oct 21, 2005 at 03:44:55PM CDT
> I believe the IP phones use regular cat5 ethernet lines, yes?  Treat them on
> the network as if they were computers connected through a hub.  You'll need
> to address the appropriate traffic to the interface on your Asterisk box.
> 
> POTS phones, of course, use single twisted pairs, and if you use cat5 you
> can get 4 phone lines in a single cat5 cable and wire them any way you wish. 
> There's no magic there.

The one caveat here is that the number of POTS phones you can drive from a
single FXS depends on the capabilities of the card or device providing the
FXS interface, and the requirements of your phones.  Each FXS has to provide
ringing voltage and current, conventional audio, and a nominal steady
voltage to the connected conventional POTS phones.  There's obviously an
electrical limit to the amount of current that the FXS interface can
provide.
 
> Thus spake Larry Alkoff on Fri, Oct 21, 2005 at 03:39:22PM CDT
> > I've just finished the O-Reilly book on Asterisk and setup a separate 
> > computer with Asterisk@Home.
> > 
> > The book has a lot of good information but for some reason does not 
> > discuss the physical wiring that goes to stations (FXS) and FXOs.
> > 
> > I understand that cat5 type wire would be used.
> > What commonly used standards, if any, are used on the various colored 
> > wires?  What signals and voltages to they carry?
> > 
> > Presently I have a lot of POTS telephones all wired in 'home run' with a 
> > single cat5 from line 1 of a station RJ-11 station block to type 66 
> > blocks where the Asterisk equipment will live.
> > 
> > Can I do the same with Asterisk station lines - that is each extension 
> > would be a single pair or two from Asterisk that carries power, ringing 
> > current and voice.
> > 
> > What equipment do I need at the Asterisk end to connect multiple analog 
> > POTS phone each station?
> > 
> > I also need to know how many wire pairs are needed for non-POTS phones 
> > and what those wire carry.
> > 
> > If feasable the easiest wiring method is to end up with all POTS phones 
> > on the default inner pair line 1 (blue) and non-POTS phones on line 2 
> > (orange) and if necessary, 3 (green) leaving 4 (extra pair possibly used 
> > for direct connection to CO line for emergencies).  That way any single 
> > (not double) line POTS phone can be safely plugged in as they are all on 
> > line 1.  Could this be done?
> > 
> > My mnemonic for telephone wire color coding is "BOG Brown".
> > 
> > I'm interested in any comments on physical wiring.
> > 
> > Larry
> > -- 
> > Larry Alkoff N2LA - Austin TX
> > Using Thunderbird on Slackware Linux
> > _______________________________________________
> > Austin-Asterisk-Users-Group mailing list
> > Austin-Asterisk-Users-Group@bybent.com
> > http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
> -- 
> Lindsay Haisley       | "Fighting against human |     PGP public key
> FMP Computer Services |    creativity is like   |      available at
> 512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
> http://www.fmp.com    |        dandelions"      |
>                       |      (Pamela Jones)     |
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 
Lindsay Haisley       | "Fighting against human |     PGP public key
FMP Computer Services |    creativity is like   |      available at
512-259-1190          |    trying to eradicate  | <http://pubkeys.fmp.com>
http://www.fmp.com    |        dandelions"      |
                      |      (Pamela Jones)     |
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Larry Alkoff wrote:

> I understand that cat5 type wire would be used.
> What commonly used standards, if any, are used on the various colored
> wires?  
For networking there are A and B color code standards, (which are marked on the
keystone connectors).  It's a little confusing when you google for  it.  Some
say use one for govt. work, the other for everything else.  Who else knows it
this has settled out to true "do it oneway" standard?

What signals and voltages to they carry?
I have not ever measured ethernet -- likely 50 to 500 millivolts differential.
POTS lines have ~48 volts usually and maybe 100 to 300 VAC when ringing.


> 
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=20

I am fairly new to asterisk and unfortunately, the only asterisk I have
time to work with and have the most success with is Asterisk@Home.

=20

I have a Digium TDM 400 card with 4 pots inputs to it.

=20

Could someone tell me how to configure Asterisk to allow one extension
or a group of extensions to use one Zap channel exclusively and another
extension or group to use a different Zap channel exclusively?

=20

I need to be able to give 4 groups access to just one Zap channel each.

=20

Thanks,

=20

I hope to see everyone at the next meeting.

=20

=20

Greg Shapiro

RF Services Inc.

7301 RR 620 North

Bldg. 155 Suite 279

Austin Texas 78726

Greg@BroadcastEngineer.com

512-692-9225

=20

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am fairly new to asterisk and unfortunately, the =
only
asterisk I have time to work with and have the most success with is
Asterisk@Home.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have a Digium TDM 400 card with 4 pots inputs to =
it.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Could someone tell me how to configure Asterisk to =
allow one
extension or a group of extensions to use one Zap channel exclusively =
and
another extension or group to use a different Zap channel =
exclusively?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I need to be able to give 4 groups access to just one =
Zap
channel each.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I hope to see everyone at the next =
meeting.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><strong><b><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Greg =
Shapiro</span></font></b></strong><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><strong><b><font
size=3D2 color=3Dmaroon face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:maroon'>RF Services Inc.</span></font></b></strong><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><strong><b><font
size=3D2 color=3Dmaroon face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:maroon'>7301 RR 620 =
North</span></font></b></strong><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><strong><b><font
size=3D2 color=3Dmaroon face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:maroon'>Bldg. 155 <st1:address w:st=3D"on"><st1:Street =
w:st=3D"on">Suite</st1:Street>
 279</st1:address></span></font></b></strong><o:p></o:p></p>

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style=3D'text-align:center'><strong><b><font
size=3D2 color=3Dmaroon face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;
color:maroon'>Austin Texas =
78726</span></font></b></strong><o:p></o:p></p>

<p class=3DMsoNormal align=3Dcenter =
style=3D'text-align:center'><strong><b><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'><a
href=3D"mailto:Greg@BroadcastEngineer.com">Greg@BroadcastEngineer.com</a>=
</span></font></b></strong><o:p></o:p></p>

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style=3D'text-align:center'><strong><b><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>512-692-9225</span></font></=
b></strong><o:p></o:p></p>

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From labradley@mindspring.com Sun Oct 23 18:47:59 2005
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Subject: [A*UG] Any nominations for a good VOIP provider?
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Looking for a good VOIP provider.  Say one with reliable service, a 24/7 
free calling plan for all US and Canada at a reasonable rate, 
cancellable without penalty and reasonably pleasant customer service.

Oh, they also have to have service and 512 area code numbers for the 
Austin area.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From carey@itfreedom.com Mon Oct 24 17:39:55 2005
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Subject: [A*UG] Asterisk virtual hosting
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Has anybody setup asterisk as true SIP virtual hosting platform before?
We're hosting multiple customers on our Asterisk platform, but right now all
users are in a 'flat' namespace.  It seems to me that it should be more like
a mail server hosting multiple virtual domains.

For example, The SIP protocol and the associated DNS-related RFCs provide a
mechanism for one SIP server to locate and connect to the telephone of
another SIP user in another domain, much like mail servers are located and
connected to in order to route email.  If I have a SIP address of
carey@fubar.com, AND my phone is hosted on a given SIP server, say,
sip.itfreedom.com (not the unrelated domain), AND I have DNS configured
properly, then another sip server should be able to:

- lookup the address of the sip server for fubar.com (i.e.,
sip.itfreedom.com)
- connect to the sip device at carey@fubar.com and start talking

I'd like to do this with Asterisk.  How?

Right now, we setup virtual sip users by creating sip.conf entries like
[carey_fubar_com], which seems kinda ugly.

I'd like to create virtual sip users like [carey@fubar.com], where fubar.com
is a CNAME to sip.itfreedom.com.

When I do that, though, Asterisk complains about a loop.  I think it tries
to look up the IP for fubar.com, routes back to itself, but can find no
[carey] user (i.e., [carey@local]), so stops.

Am I making any sense?

Carey

From aaug.voip.taylorcc@dfgh.net Mon Oct 24 22:12:38 2005
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Chris Tooley - ctooley@gnumber.com wrote:
> Digium has great support for their cards.  If you call them, they will
> resolve these issues for even ssh'ing to the machine and connecting to a
> shared screen session so that you can see what they're doing.
> 

I recommend turning on recording in the screen session.  Then you can use it 
as reference later.

From aaug.voip.taylorcc@dfgh.net Mon Oct 24 23:16:22 2005
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Larry Alkoff - labradley@mindspring.com wrote:
> Looking for a good VOIP provider.  Say one with reliable service, a 24/7 
> free calling plan for all US and Canada at a reasonable rate, 
> cancellable without penalty and reasonably pleasant customer service.
> 
> Oh, they also have to have service and 512 area code numbers for the 
> Austin area.

The more customer oriented you get the less configurable from what I have 
seen.. Ie. the more you should be technical.  On the voip-info.org voip 
provider list you can start checking out various providers and the other voip 
forums for reviews.

Telasip seems to be the most flexible for BYOD for a unlimited plan.  After 
that I would consider SunRocket... and at some point FWD (when they get there 
out bound plan moving).

BroadVoice customer service is terrible... but for $10/month instate or less 
for inbound DID only they would make a OK secondary provider.  Of course for 
backup purposes I would probably rather go with Voxee.com at 1.1c/min or 
possibly Jajah.com for less (which also connects to skype).

Taylor
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Here is a question I tried to pose on the IRC line:


Question: I have an asterisk system with a Digium TDM02B: TDM400P + 2  
PORT FXO
it does not seem to be able to complete calls.  I.e., I get an  
operator error message that the call cannot complete as dialed.

when I look at the call log on the AMP, it shows the correct  
numbers.  Thus, the telephone sets are transmitting correctly to the  
the Asterisk box.  but from there something goes wrong.  any thoughts  
on how to debug that?

the odd thing is that some numbers work, others don't.  to make  
things simple I set up to have only one outbound route.  A telephone  
connected directly to the PSTN has no trouble with the same numbers.   
And, oddly, this all seemed to work fine a few days ago.

Does anyone on this forum have some thoughts on that?



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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">Here is a question I tried to =
pose on the IRC line:<DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV=
 style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" color=3D"#888888" =
face=3D"Lucida Grande" size=3D"1"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 9px;"><BR =
class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;">Question: I have an asterisk system with a Digium TDM02B: TDM400P =
+ 2 PORT FXO</SPAN></FONT></DIV><DIV style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Lucida Grande" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 11px;">it does not seem =
to be able to complete calls.=A0 I.e., I get an operator error message =
that the call cannot complete as dialed.</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" color=3D"#888888" =
face=3D"Lucida Grande" size=3D"1"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 9px;"><BR =
class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;">when I look at the call log on the AMP, it shows the correct =
numbers.=A0 Thus, the telephone sets are transmitting correctly to the =
the Asterisk box.=A0 but from there something goes wrong.=A0 any =
thoughts on how to debug that?</SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;"><BR class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;">the odd thing is that some numbers work, others don't.=A0 to make =
things simple I set up to have only one outbound route.=A0 A telephone =
connected directly to the PSTN has no trouble with the same numbers.=A0 =
And, oddly, this all seemed to work fine a few days =
ago.</SPAN></FONT></DIV><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Lucida Grande" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 11px;"><BR =
class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;">Does anyone on this forum have some thoughts on =
that?</SPAN></FONT></DIV><DIV style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; "><FONT =
class=3D"Apple-style-span" face=3D"Lucida Grande" size=3D"3"><SPAN =
class=3D"Apple-style-span" style=3D"font-size: 11px;"><BR =
class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" face=3D"Lucida =
Grande" size=3D"3"><SPAN class=3D"Apple-style-span" style=3D"font-size: =
11px;"><BR class=3D"khtml-block-placeholder"></SPAN></FONT></DIV><DIV =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><FONT class=3D"Apple-style-span" color=3D"#888888" =
face=3D"Lucida Grande" size=3D"1"><SPAN class=3D"Apple-style-span" =
style=3D"font-size: 9px;"></SPAN></FONT></DIV></BODY></HTML>=

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From fc4linux@kojetin.org Thu Oct 27 00:07:42 2005
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Subject: [A*UG] BEST VOIP PHONE
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I have read over many many product specification from various vender's 
websites. The one thing I can't find much of are user reviews. Much less 
reviews from users that happen to have an Asterisk setup in their home.

Here is what I would like:

- A regular voip phone at my desk such as the Cisco 7960. However, those 
Cisco 7960's are pretty expensive for what I will be using the phone 
for.  I have looked at a couple of SNOM voip phones but then read that 
the voice quality on those phones are similar to a USB phone. In other 
words, horrible.

- I also want an 802.11b/g wifi voip phone that I can leave in the 
family room. I am just not sure about the voice quality and range on 
them though.

If anyone has any particular experiences with some good phones; please 
help me choose one! There are so many!


Thanks,

Mike
From ctooley@gnumber.com Thu Oct 27 09:21:48 2005
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Subject: Re: [A*UG] BEST VOIP PHONE
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For a deskphone I highly recommend the newer Polycom phones.  They're
far less expensive than the Cisco phones but Cisco, Polycom and a
company called Shoreline jointly develop the phones so they are in
effect the same as the Cisco phones in a different skin.  They're
nowhere near as expenisve as the Ciscos.

For a cordless phone I recommend the Pulver model.  Remember though that
802.11 phones are going to be expensive and have a the problem that if
your wireless service weakens the call just drops, it doesn't just get
fuzzy.

On Thu, 2005-10-27 at 00:07 -0500, Michael Kojetin wrote:
> I have read over many many product specification from various vender's 
> websites. The one thing I can't find much of are user reviews. Much less 
> reviews from users that happen to have an Asterisk setup in their home.
> 
> Here is what I would like:
> 
> - A regular voip phone at my desk such as the Cisco 7960. However, those 
> Cisco 7960's are pretty expensive for what I will be using the phone 
> for.  I have looked at a couple of SNOM voip phones but then read that 
> the voice quality on those phones are similar to a USB phone. In other 
> words, horrible.
> 
> - I also want an 802.11b/g wifi voip phone that I can leave in the 
> family room. I am just not sure about the voice quality and range on 
> them though.
> 
> If anyone has any particular experiences with some good phones; please 
> help me choose one! There are so many!
> 
> 
> Thanks,
> 
> Mike
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From carey@itfreedom.com Thu Oct 27 10:24:54 2005
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> 
> For a deskphone I highly recommend the newer Polycom phones.
>

I agree on that.  We're deploying them exclusively

> 
> For a cordless phone I recommend the Pulver model.

Haven't tried that one.  We're playing with a Hitachi-Cable phone that's
really nice -- looks like a cell phone.  It's current limitation is no WPA
encryption; it's WEP only.  I'm not sure if there's a firmware upgrade
available for it.  What kind of encryption does the Pulver phone support?

Carey

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> > For a cordless phone I recommend the Pulver model.
> 
> Haven't tried that one.  We're playing with a Hitachi-Cable phone that's
> really nice -- looks like a cell phone.  It's current limitation is no WPA
> encryption; it's WEP only.  I'm not sure if there's a firmware upgrade
> available for it.  What kind of encryption does the Pulver phone support?

The old one from pulver did WEP only.  I don't know about new ones.
They're at pulverinnovations.com

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Subject: [A*UG] Shared registrations?
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While we're chatting about Polycoms, does anybody have shared registrations
working with Asterisk and Polycom's, in particular.  That's where two phones
register as the same user and behave as an old-fashioned party line, in
essence.
 
- calls ringing their extension ring both phones
- either phone can see the status of the 'line', ie, whether someone's on
the line, away, etc.
- either phone can pick up and join a conversation.
 
Thanks in advance,
Carey
 
  _____  

Carey Jung
IT FreedomR
 <mailto:carey@itfreedom.com> carey@itfreedom.com
512.419.0070, fax 419.0080
 
 

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<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial size=3D2>While =
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calls ringing=20
their extension ring both phones</FONT></SPAN></DIV>
<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial size=3D2>- =
either phone can=20
see the status of the 'line', ie, whether someone's on the line, away,=20
etc.</FONT></SPAN></DIV>
<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial size=3D2>- =
either phone can=20
pick up and join a conversation.</FONT></SPAN></DIV>
<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial size=3D2>Thanks =
in=20
advance,</FONT></SPAN></DIV>
<DIV><SPAN class=3D096044515-27102005><FONT face=3DArial=20
size=3D2>Carey</FONT></SPAN></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV><FONT face=3DArial =
size=3D2>
<DIV>
<HR color=3D#808080>
</DIV>
<DIV align=3Dright><FONT color=3D#808080 size=3D1><SPAN =
class=3D290161905-18101999>Carey=20
Jung</SPAN></FONT></DIV>
<DIV align=3Dright><FONT size=3D1><FONT face=3D"Arial Black"><SPAN=20
class=3D290161905-18101999><EM><FONT color=3D#ff0000>IT </FONT><FONT=20
color=3D#0000ff>Freedom&reg;</FONT></EM></SPAN></FONT></FONT></DIV>
<DIV align=3Dright><FONT size=3D1><FONT face=3DArial><SPAN =
class=3D290161905-18101999><A=20
href=3D"mailto:carey@itfreedom.com"><FONT=20
color=3D#808080>carey@itfreedom.com</FONT></A><FONT face=3D"Arial Black" =

color=3D#0000ff><SPAN=20
class=3D420563805-18101999><EM></EM></SPAN></FONT></SPAN></FONT></FONT></=
DIV>
<DIV align=3Dright><FONT face=3DArial color=3D#808080 size=3D1><SPAN=20
class=3D290161905-18101999>512.419.0070, fax =
419.0080</SPAN></FONT></DIV></FONT>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_005E_01C5DAE3.E3CD4880--

From gil@kloepfer.org Thu Oct 27 11:32:00 2005
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Date: Thu, 27 Oct 2005 10:56:50 -0500
From: Gil Kloepfer <aaug@kloepfer.org>
To: austin-asterisk-users-group@bybent.com
Subject: Re: [A*UG] Shared registrations?
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On Thu, Oct 27, 2005 at 10:47:56AM -0500, Carey Jung wrote:
> While we're chatting about Polycoms, does anybody have shared registrations
> working with Asterisk and Polycom's, in particular.  That's where two phones
> register as the same user and behave as an old-fashioned party line, in
> essence.
>  
> - calls ringing their extension ring both phones
> - either phone can see the status of the 'line', ie, whether someone's on
> the line, away, etc.
> - either phone can pick up and join a conversation.

Shared registrations don't work in Asterisk currently (including -head)
because the event type that Polycom uses is not supported in Asterisk.

The dialog event type IS supported in Asterisk (Polycom says it
uses this for "bridged" lines), however I have not been able to get the
Polycom phone to subscribe to this event type.

There is an "IM buddy" hack (see voip-info.org) that can be used to
get the status of another extension.  However, this will not allow
both phones to ring at the same time.

You can make both phones ring at the same time by using "&" and two
or more tech/resource designators in the Dial application.  However,
this doesn't connect into the "buddy" thing at all (you'd need to
either just ring both phones, and you wouldn't know which extension
was really being called, or you'd need another line button for a
pseudo-extension on the second phone).

Picking-up and joining a conversation is actually a rather complex
thing as far as SIP is concerned.  Remember that in networking these
are point-to-point connections and in order to have more than one
member of a conversation, one party (or the PBX) needs to mix the
conversation together.  I don't know currently of anything that can
do this on-the-fly.

---
Gil Kloepfer
aaug@kloepfer.org
From carey@itfreedom.com Thu Oct 27 11:40:02 2005
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> 
> There is an "IM buddy" hack (see voip-info.org) that can be used to
> get the status of another extension.  However, this will not allow
> both phones to ring at the same time.

Saw that.  Haven't had time to play with it, though.

> 
> You can make both phones ring at the same time by using "&" and two
> or more tech/resource designators in the Dial application.  However,
> this doesn't connect into the "buddy" thing at all (you'd need to
> either just ring both phones, and you wouldn't know which extension
> was really being called, or you'd need another line button for a
> pseudo-extension on the second phone).
> 
> Picking-up and joining a conversation is actually a rather complex
> thing as far as SIP is concerned.  Remember that in networking these
> are point-to-point connections and in order to have more than one
> member of a conversation, one party (or the PBX) needs to mix the
> conversation together.  I don't know currently of anything that can
> do this on-the-fly.

The only thing that came to my mind was perhaps some trickery with meetme.

Carey

From Greg@towerz.net Thu Oct 27 12:20:28 2005
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I have been using the ATCOM phones using IAX2 and find that they work
the best of any SIP phones ever tried. I also have the Pulver WIFI phone
but they aren't made anymore. It works OK but has registration problems
staying connected and registered.

IAX2 is the best protocol for getting through firewalls.

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225
 

-----Original Message-----
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than "Re: Contents of Austin-Asterisk-Users-Group digest..."


Today's Topics:

   1. BEST VOIP PHONE (Michael Kojetin)
   2. Re: BEST VOIP PHONE (Chris Tooley)
   3. RE: BEST VOIP PHONE (Carey Jung)
   4. RE: BEST VOIP PHONE (Chris Tooley)
   5. Shared registrations? (Carey Jung)
   6. Re: Shared registrations? (Gil Kloepfer)
   7. RE: Shared registrations? (Carey Jung)


----------------------------------------------------------------------

Message: 1
Date: Thu, 27 Oct 2005 00:07:37 -0500
From: Michael Kojetin <fc4linux@kojetin.org>
Subject: [A*UG] BEST VOIP PHONE
To: "[A*UG]" <austin-asterisk-users-group@bybent.com>
Message-ID: <43606099.9030702@kojetin.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

I have read over many many product specification from various vender's 
websites. The one thing I can't find much of are user reviews. Much less

reviews from users that happen to have an Asterisk setup in their home.

Here is what I would like:

- A regular voip phone at my desk such as the Cisco 7960. However, those

Cisco 7960's are pretty expensive for what I will be using the phone 
for.  I have looked at a couple of SNOM voip phones but then read that 
the voice quality on those phones are similar to a USB phone. In other 
words, horrible.

- I also want an 802.11b/g wifi voip phone that I can leave in the 
family room. I am just not sure about the voice quality and range on 
them though.

If anyone has any particular experiences with some good phones; please 
help me choose one! There are so many!


Thanks,

Mike

------------------------------

Message: 2
Date: Thu, 27 Oct 2005 09:21:47 -0500
From: Chris Tooley <ctooley@gnumber.com>
Subject: Re: [A*UG] BEST VOIP PHONE
To: "[A*UG]" <austin-asterisk-users-group@bybent.com>
Message-ID: <1130422907.22671.12.camel@localhost.localdomain>
Content-Type: text/plain

For a deskphone I highly recommend the newer Polycom phones.  They're
far less expensive than the Cisco phones but Cisco, Polycom and a
company called Shoreline jointly develop the phones so they are in
effect the same as the Cisco phones in a different skin.  They're
nowhere near as expenisve as the Ciscos.

For a cordless phone I recommend the Pulver model.  Remember though that
802.11 phones are going to be expensive and have a the problem that if
your wireless service weakens the call just drops, it doesn't just get
fuzzy.

On Thu, 2005-10-27 at 00:07 -0500, Michael Kojetin wrote:
> I have read over many many product specification from various vender's

> websites. The one thing I can't find much of are user reviews. Much
less 
> reviews from users that happen to have an Asterisk setup in their
home.
> 
> Here is what I would like:
> 
> - A regular voip phone at my desk such as the Cisco 7960. However,
those 
> Cisco 7960's are pretty expensive for what I will be using the phone 
> for.  I have looked at a couple of SNOM voip phones but then read that

> the voice quality on those phones are similar to a USB phone. In other

> words, horrible.
> 
> - I also want an 802.11b/g wifi voip phone that I can leave in the 
> family room. I am just not sure about the voice quality and range on 
> them though.
> 
> If anyone has any particular experiences with some good phones; please

> help me choose one! There are so many!
> 
> 
> Thanks,
> 
> Mike
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com


------------------------------

Message: 3
Date: Thu, 27 Oct 2005 10:24:47 -0500
From: "Carey Jung" <carey@itfreedom.com>
Subject: RE: [A*UG] BEST VOIP PHONE
To: "'Chris Tooley'" <ctooley@gnumber.com>,   "'[A*UG]'"
	<austin-asterisk-users-group@bybent.com>
Message-ID: <200510271524.j9RFOrs7010659@buzzard.onr.com>
Content-Type: text/plain;	charset="us-ascii"

> 
> For a deskphone I highly recommend the newer Polycom phones.
>

I agree on that.  We're deploying them exclusively

> 
> For a cordless phone I recommend the Pulver model.

Haven't tried that one.  We're playing with a Hitachi-Cable phone that's
really nice -- looks like a cell phone.  It's current limitation is no
WPA
encryption; it's WEP only.  I'm not sure if there's a firmware upgrade
available for it.  What kind of encryption does the Pulver phone
support?

Carey


------------------------------

Message: 4
Date: Thu, 27 Oct 2005 10:33:41 -0500
From: Chris Tooley <chris@tooley.com>
Subject: RE: [A*UG] BEST VOIP PHONE
To: Carey Jung <carey@itfreedom.com>
Cc: "'\[A*UG\]'" <austin-asterisk-users-group@bybent.com>
Message-ID: <1130427221.22671.19.camel@localhost.localdomain>
Content-Type: text/plain


> > For a cordless phone I recommend the Pulver model.
> 
> Haven't tried that one.  We're playing with a Hitachi-Cable phone
that's
> really nice -- looks like a cell phone.  It's current limitation is no
WPA
> encryption; it's WEP only.  I'm not sure if there's a firmware upgrade
> available for it.  What kind of encryption does the Pulver phone
support?

The old one from pulver did WEP only.  I don't know about new ones.
They're at pulverinnovations.com


------------------------------

Message: 5
Date: Thu, 27 Oct 2005 10:47:56 -0500
From: "Carey Jung" <carey@itfreedom.com>
Subject: [A*UG] Shared registrations?
To: "'[A*UG]'" <austin-asterisk-users-group@bybent.com>
Message-ID: <200510271548.j9RFm0s7011297@buzzard.onr.com>
Content-Type: text/plain; charset="us-ascii"

While we're chatting about Polycoms, does anybody have shared
registrations
working with Asterisk and Polycom's, in particular.  That's where two
phones
register as the same user and behave as an old-fashioned party line, in
essence.
 
- calls ringing their extension ring both phones
- either phone can see the status of the 'line', ie, whether someone's
on
the line, away, etc.
- either phone can pick up and join a conversation.
 
Thanks in advance,
Carey
 
  _____  

Carey Jung
IT FreedomR
 <mailto:carey@itfreedom.com> carey@itfreedom.com
512.419.0070, fax 419.0080
 
 
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------------------------------

Message: 6
Date: Thu, 27 Oct 2005 10:56:50 -0500
From: Gil Kloepfer <aaug@kloepfer.org>
Subject: Re: [A*UG] Shared registrations?
To: austin-asterisk-users-group@bybent.com
Message-ID: <20051027155650.GA20536@kloepfer.org>
Content-Type: text/plain; charset=us-ascii

On Thu, Oct 27, 2005 at 10:47:56AM -0500, Carey Jung wrote:
> While we're chatting about Polycoms, does anybody have shared
registrations
> working with Asterisk and Polycom's, in particular.  That's where two
phones
> register as the same user and behave as an old-fashioned party line,
in
> essence.
>  
> - calls ringing their extension ring both phones
> - either phone can see the status of the 'line', ie, whether someone's
on
> the line, away, etc.
> - either phone can pick up and join a conversation.

Shared registrations don't work in Asterisk currently (including -head)
because the event type that Polycom uses is not supported in Asterisk.

The dialog event type IS supported in Asterisk (Polycom says it
uses this for "bridged" lines), however I have not been able to get the
Polycom phone to subscribe to this event type.

There is an "IM buddy" hack (see voip-info.org) that can be used to
get the status of another extension.  However, this will not allow
both phones to ring at the same time.

You can make both phones ring at the same time by using "&" and two
or more tech/resource designators in the Dial application.  However,
this doesn't connect into the "buddy" thing at all (you'd need to
either just ring both phones, and you wouldn't know which extension
was really being called, or you'd need another line button for a
pseudo-extension on the second phone).

Picking-up and joining a conversation is actually a rather complex
thing as far as SIP is concerned.  Remember that in networking these
are point-to-point connections and in order to have more than one
member of a conversation, one party (or the PBX) needs to mix the
conversation together.  I don't know currently of anything that can
do this on-the-fly.

---
Gil Kloepfer
aaug@kloepfer.org

------------------------------

Message: 7
Date: Thu, 27 Oct 2005 11:39:58 -0500
From: "Carey Jung" <carey@itfreedom.com>
Subject: RE: [A*UG] Shared registrations?
To: "'Gil Kloepfer'" <aaug@kloepfer.org>,
	austin-asterisk-users-group@bybent.com
Message-ID: <200510271640.j9RGe2s7013093@buzzard.onr.com>
Content-Type: text/plain;	charset="us-ascii"

> 
> There is an "IM buddy" hack (see voip-info.org) that can be used to
> get the status of another extension.  However, this will not allow
> both phones to ring at the same time.

Saw that.  Haven't had time to play with it, though.

> 
> You can make both phones ring at the same time by using "&" and two
> or more tech/resource designators in the Dial application.  However,
> this doesn't connect into the "buddy" thing at all (you'd need to
> either just ring both phones, and you wouldn't know which extension
> was really being called, or you'd need another line button for a
> pseudo-extension on the second phone).
> 
> Picking-up and joining a conversation is actually a rather complex
> thing as far as SIP is concerned.  Remember that in networking these
> are point-to-point connections and in order to have more than one
> member of a conversation, one party (or the PBX) needs to mix the
> conversation together.  I don't know currently of anything that can
> do this on-the-fly.

The only thing that came to my mind was perhaps some trickery with
meetme.

Carey


------------------------------

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Austin-Asterisk-Users-Group@bybent.com
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From: Willie McKemie <sprinter@austinfarm.org>
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Subject: [A*UG] New asterisk@home install
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I bought a couple of Grandstream 100s that seem to work well.  Except 
for echoing on "speaker".  I got asterisk@home 1.5 installed without 
incident.  I seem to have successfully set up asterisk to recognize 
the Grandstreams as extensions.  I have seem to have asterisk answering 
a plain old telephone line (via a Digium clone card).

Here are my current "hang ups" I hope someone here will help with:

1) I haven't figured out how to configure the outgoing POTS stuff.  All 
the prefix conversion stuff seem like gobbledgoop to me.  So far.  Can 
someone enlighten me?  Or, better yet, give me a recipe?

2) I can't receive voicemail.  I can leave a message from one extension 
to another if the target extension is busy or does not answer.  
However, when I try to receive the message and the "operator" tells me 
to "press one to....", keyins have no effect.

3) I have setup both extensions to be on a "ring group" with the hope 
that an incoming call would ring on BOTH extensions.  Seemingling not 
so, only the first extension in the "ring group" seems to ring.  Can 
someone suggest how to configure so that it does what I wish?  When I 
get additional extensions, I will also want to have one extension ring 
ALL the others.

I have two phone lines; my intermediate term goal is to add a second 
Digium clone for the other phone line and configure so that any 
extension can dial out on either line.  Something like 
"8+<7or10or11digits>" for one line and "9+<7or10or11digits>" for the 
other line.  I presume that that is doable?  Currently, one line is a 
"metro" line, which can call Austin without extra charge, and one is 
not.

My longer term goal is to get internet VOIP working acceptable well 
over my satellite connection and then drop one of my land lines.  My 
expectations here are rather low due the troublesomeness of my Direcway 
provider.

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  489 days 22 hours 55 minutes
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Date: Sun, 30 Oct 2005 14:06:05 -0600
From: Willie McKemie <sprinter@austinfarm.org>
To: austin-asterisk-users-group@bybent.com
Subject: Re: [A*UG] New asterisk@home install
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On Sun, Oct 30, 2005 at 01:31:00PM -0600, Michael Kojetin wrote:
> >1) I haven't figured out how to configure the outgoing POTS stuff.  All 
> >the prefix conversion stuff seem like gobbledgoop to me.  So far.  Can 
> >someone enlighten me?  Or, better yet, give me a recipe?
> > 
> >
> What type of prefix conversion....     are you talking about the  
> _NXXNXXXXXX   type stuff?

Yes.  I'm a total newbie at this telephone stuff.

> 
> What kind of Digium clone card do you have? The only clones I have seen 
> on Ebay just had a single FXO and no FXS.  How much did you pay for it?

Yeah, FXO only; two RJ11 phone jacks on it.  I bought two to average 
the shipping down; about $25 all together, as I recall.

Asterisk seems a bit sluggish answering the phone line as well as 
dealing with extension to extension calls; incoming calls ring twice 
before Asterisk takes over.  I wonder if that is a result of running 
on a 64mb, 330mhz box?  Probably the 64mb.  I see it setup a 800mb 
swap and is using a little of it.  I think I'll look around for a bit 
more memory for it.

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  490 days  3 hours 03 minutes
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Subject: Re: [A*UG] New asterisk@home install
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On Sun, Oct 30, 2005 at 02:06:05PM -0600, Willie McKemie wrote:
> Asterisk seems a bit sluggish answering the phone line as well as 
> dealing with extension to extension calls; incoming calls ring twice 
> before Asterisk takes over.  I wonder if that is a result of running 
> on a 64mb, 330mhz box?  Probably the 64mb.  I see it setup a 800mb 
> swap and is using a little of it.  I think I'll look around for a bit 
> more memory for it.

Actually, the reason Asterisk answers on the second ring is because
Caller-ID information is sent by the telco as a burst of FSK data
between the first and second ring.  If it answered on the first
ring, then the Caller-ID information would be lost.

I suspect that if you weren't interested in the Caller-ID info, you
could set "usecallerid=no" in zapata.conf and it would begin answering
on the first ring.

---
Gil Kloepfer
aaug@kloepfer.org
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On Sun, Oct 30, 2005 at 02:22:15PM -0600, Gil Kloepfer wrote (without
thinking):
> Actually, the reason Asterisk answers on the second ring is because
[...]

And by "Asterisk answers" I meant "Asterisk acts on the call."  The
call isn't actually "answered" until the remote side picks-up.

Argh.  It's a Sunday and my brain isn't all there...

By the way, if you use a Sipura SPA-3000 (with the port that can
connect to a phone line), it actually DOES *answer* the phone line
when sending the call to the PBX.  Sigh.  Hopefully they (or Cisco)
fix or have fixed that in subsequent versions of that box.

---
Gil Kloepfer
aaug@kloepfer.org
From sprinter@austinfarm.org Sun Oct 30 14:43:15 2005
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On Sun, Oct 30, 2005 at 02:22:15PM -0600, Gil Kloepfer wrote:
> On Sun, Oct 30, 2005 at 02:06:05PM -0600, Willie McKemie wrote:
> > Asterisk seems a bit sluggish answering the phone line as well as 
> > dealing with extension to extension calls; incoming calls ring twice 
> > before Asterisk takes over.  I wonder if that is a result of running 
> > on a 64mb, 330mhz box?  Probably the 64mb.  I see it setup a 800mb 
> > swap and is using a little of it.  I think I'll look around for a bit 
> > more memory for it.
> 
> Actually, the reason Asterisk answers on the second ring is because
> Caller-ID information is sent by the telco as a burst of FSK data
> between the first and second ring.  If it answered on the first
> ring, then the Caller-ID information would be lost.
> 
> I suspect that if you weren't interested in the Caller-ID info, you
> could set "usecallerid=no" in zapata.conf and it would begin answering
> on the first ring.

I suspect that, since I'm not paying the telco for it, caller-id info 
is not there.  You reckon that's so?

Thanks for the information!

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  490 days  3 hours 35 minutes
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On Oct 30, 2005, at 10:27 AM, Willie McKemie wrote:


> I bought a couple of Grandstream 100s that seem to work well.  Except
> for echoing on "speaker".  I got asterisk@home 1.5 installed without
> incident.  I seem to have successfully set up asterisk to recognize
> the Grandstreams as extensions.  I have seem to have asterisk  
> answering
> a plain old telephone line (via a Digium clone card).
>

We also have a couple of Grandstream phones (Budge Tone - I don't  
recall if that is the same as the 100s).  We are ready to give up on  
them because of outgoing voice quality issues - it is getting  
tiresome to have people say "you sound muffled".  It seems that the  
music on hold is better, so I am suspecting the phone itself.  We  
also used the clone cards for about a week and went to the Digium  
card.  However, I am not sure if the clone card was the problem.

>
> Here are my current "hang ups" I hope someone here will help with:
>
> 1) I haven't figured out how to configure the outgoing POTS stuff.   
> All
> the prefix conversion stuff seem like gobbledgoop to me.  So far.  Can
> someone enlighten me?  Or, better yet, give me a recipe?
>
> 2) I can't receive voicemail.  I can leave a message from one  
> extension
> to another if the target extension is busy or does not answer.
> However, when I try to receive the message and the "operator" tells me
> to "press one to....", keyins have no effect.
>

You probably have the DTMF set differently on the phone from  
Asterisk.  Probably it is set to RFC2833 in asterisk and something  
else on the phone.   Go to the setup for the phone and check it (on  
mine it is a field called "Send DTMF").  Then change it to be the  
same as in Asterisk (check extensions in AMP).  I read somewhere  
(Hitchhiker's Guide to Asterisk - http://www.asteriskdocs.org/modules/ 
tinycontent/content/docbook/current/docs-html/book1.html) that Budge  
Tone phones require DTMF to be SIP-INFO.  However, that is not true.   
Could be that is the case with older firmware for the phone because  
there must be some basis for that assertion.  You may want to make  
sure you have the latest firmware for the phone.

>
> 3) I have setup both extensions to be on a "ring group" with the hope
> that an incoming call would ring on BOTH extensions.  Seemingling not
> so, only the first extension in the "ring group" seems to ring.  Can
> someone suggest how to configure so that it does what I wish?  When I
> get additional extensions, I will also want to have one extension ring
> ALL the others.
>
> I have two phone lines; my intermediate term goal is to add a second
> Digium clone for the other phone line and configure so that any
> extension can dial out on either line.  Something like
> "8+<7or10or11digits>" for one line and "9+<7or10or11digits>" for the
> other line.  I presume that that is doable?  Currently, one line is a
> "metro" line, which can call Austin without extra charge, and one is
> not.
>
> My longer term goal is to get internet VOIP working acceptable well
> over my satellite connection and then drop one of my land lines.  My
> expectations here are rather low due the troublesomeness of my  
> Direcway
> provider.
>
> -- 
> Willie, ONWARD!  Through the fog!
> http://counter.li.org Linux registered user #228836 since 1995
> Linux system uptime  489 days 22 hours 55 minutes
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
>
>


From gil@kloepfer.org Wed Nov  2 11:19:35 2005
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Just wanted to pass along to the folks here that at the next
meeting of the Capital Area Central Texas UNIX Society (CACTUS)
we will be hosting an "installfest" -- people are invited to
bring equipment to the meeting where (hopefully) we can install
operating systems and open source software in the company of
experienced users.

I'm planning on being there to help set-up Asterisk.  So if anyone
is interested in coming down to get a basic install going, feel
free to join us.  I'd also be happy to try to answer any of the harder
questions that come-up (I've done presentations on Asterisk at
CACTUS before, and am working on a large implementation where I work).

The meeting is Thursday, November 17.  Information about CACTUS,
including meeting times, directions, and specifics about our group
is available at http://www.cactus.org/.  The latest newsletter is
not yet available, but there is mention of the installfest in
the October newsletter at:

     http://www.cactus.org/CACTUS/Newsletter/cn200510.html

Please be sure to note that attendees should bring their own network
cables (long ones, at least 15 ft. (5 meters)), power strip, and
anything else they need to get their computer operational as we will
have limited resources and time.  Also note that while we have
pizza at the meetings, we ask that those who partake contribute $3
so that we don't deplete our monetary resources (we aren't a
pizza-eating organization).

Hope to see you there!

---
Gil Kloepfer
aaug@kloepfer.org
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From: Willie McKemie <sprinter@spamcop.net>
To: austin-asterisk-users-group@bybent.com
Subject: Re: [A*UG] New asterisk@home install
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> > We also have a couple of Grandstream phones (Budge Tone - I don't  
> > recall if that is the same as the 100s).  We are ready to give up on  
> > them because of outgoing voice quality issues - it is getting  
> > tiresome to have people say "you sound muffled".  It seems that the  
> > music on hold is better, so I am suspecting the phone itself.  We  
> > also used the clone cards for about a week and went to the Digium  
> > card.  However, I am not sure if the clone card was the problem.
> 
> My Grandstreams (Budge Tone-100) seem to work well on my lan.  Do you 
> want to sell your Grandstreams?

I finally got my AAH setup semi-working with POTS incoming and outgoing 
calls.  For outgoing calls (not much testing on incoming) with my 
Grandstream BT-100, I hear a slight echo of my own voice and folks on 
the other end report that I sound "far away".  SO!  I am interested in 
what is needed to correct the problem: a better IP phone, a real Digium 
card, or both.
> > 
> > You probably have the DTMF set differently on the phone from  
> > Asterisk.  Probably it is set to RFC2833 in asterisk and something  
> > else on the phone.   Go to the setup for the phone and check it (on  
> > mine it is a field called "Send DTMF").  Then change it to be the  
> > same as in Asterisk (check extensions in AMP).  I read somewhere  
> > (Hitchhiker's Guide to Asterisk - http://www.asteriskdocs.org/modules/ 
> > tinycontent/content/docbook/current/docs-html/book1.html) that Budge  
> > Tone phones require DTMF to be SIP-INFO.  However, that is not true.   
> > Could be that is the case with older firmware for the phone because  
> > there must be some basis for that assertion.  You may want to make  
> > sure you have the latest firmware for the phone.
> 
> Thanks for the pointer and the starting point; I got my Grandstreams to 
> work by configuring "Send DTMF" to "via SIP INFO".  At least, message 
> retrieval now works.

I spoke too soon.  It seems voice mail sometimes recognizes my keytones 
and sometimes not.

I notice I get email when a voice mail message is left.  And that the 
messages are available through the Asterisk web server.  That would be 
a really neat feature for travel.  If I could get through my firewall.

-- 
Willie, ONWARD!  Through the fog!
http://counter.li.org Linux registered user #228836 since 1995
Linux system uptime  18 hours 25 minutes
From fc4linux@kojetin.org Mon Nov  7 10:45:05 2005
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Willie McKemie wrote:

>>>We also have a couple of Grandstream phones (Budge Tone - I don't  
>>>recall if that is the same as the 100s).  We are ready to give up on  
>>>them because of outgoing voice quality issues - it is getting  
>>>tiresome to have people say "you sound muffled".  It seems that the  
>>>music on hold is better, so I am suspecting the phone itself.  We  
>>>also used the clone cards for about a week and went to the Digium  
>>>card.  However, I am not sure if the clone card was the problem.
>>>      
>>>
>>My Grandstreams (Budge Tone-100) seem to work well on my lan.  Do you 
>>want to sell your Grandstreams?
>>    
>>
>
>I finally got my AAH setup semi-working with POTS incoming and outgoing 
>calls.  For outgoing calls (not much testing on incoming) with my 
>Grandstream BT-100, I hear a slight echo of my own voice and folks on 
>the other end report that I sound "far away".  SO!  I am interested in 
>what is needed to correct the problem: a better IP phone, a real Digium 
>card, or both.
>  
>
>>>You probably have the DTMF set differently on the phone from  
>>>Asterisk.  Probably it is set to RFC2833 in asterisk and something  
>>>else on the phone.   Go to the setup for the phone and check it (on  
>>>mine it is a field called "Send DTMF").  Then change it to be the  
>>>same as in Asterisk (check extensions in AMP).  I read somewhere  
>>>(Hitchhiker's Guide to Asterisk - http://www.asteriskdocs.org/modules/ 
>>>tinycontent/content/docbook/current/docs-html/book1.html) that Budge  
>>>Tone phones require DTMF to be SIP-INFO.  However, that is not true.   
>>>Could be that is the case with older firmware for the phone because  
>>>there must be some basis for that assertion.  You may want to make  
>>>sure you have the latest firmware for the phone.
>>>      
>>>
>>Thanks for the pointer and the starting point; I got my Grandstreams to 
>>work by configuring "Send DTMF" to "via SIP INFO".  At least, message 
>>retrieval now works.
>>    
>>
>
>I spoke too soon.  It seems voice mail sometimes recognizes my keytones 
>and sometimes not.
>
>I notice I get email when a voice mail message is left.  And that the 
>messages are available through the Asterisk web server.  That would be 
>a really neat feature for travel.  If I could get through my firewall.
>
>  
>
Willie,

I am not sure if you read the Linux Journal, but there was a great 
article about echo cancellation in the October issue. I believe it was 
the October, maybe November issue. It was the issue with the darn 
Pinball machine makeover.

Mike
From ted.a.wolf@hp.com Thu Nov  3 17:16:30 2005
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Thread-Topic: Looking for a Sr. Networking Engineer
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From: "Wolf, Ted (ISS/Austin TX)" <ted.a.wolf@hp.com>
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HP's Network System Products organization is seeking a networking expert
for our Product Assurance Team. This position will support the complete
line of ProLiant networking products including NICs, Switches and our
sophisticated software stack.

Job Description: A Senior Test Engineer will apply advanced subject
matter knowledge of networking and switching technology to business
issues.   A Senior Test Engineer will frequently contribute to the
development of new ideas and methods for solving complex technical
problems. In addition, this individual will work on complex
problems/projects where analysis of situations or data requires an
in-depth evaluation of multiple factors. 

A Senior Test Engineer must be capable of exercising significant
independent judgment within broadly defined policies and practices to
determine best method for accomplishing work and achieving objectives.
This Engineer will lead and provide expertise to functional project
teams and may participate in cross-functional initiatives. 

A Senior Test Engineer is expected to provide mentoring and guidance to
lower level employees. The candidate frequently represents the
organization to external customers/clients. Occasional travel may be
required.

Responsibilities: 

*	Developing and Reviewing  technical documentation associated
with NSP programs
*	Analyzing  product specifications and defining test requirements
*	Developing test procedures, test programs and other tools
*	Executing tests against both ISS and NSP Products
*	Analyzing the capabilities of potential partners and vendors
*	Developing and delivering technical presentations and
demonstrations
*	Training and mentoring Test Engineers, Support Engineers and
Development Engineers

Expectations: 

*	This Engineer will be a technical advisor to NSP's Management
Staff
*	This Engineer will define new and novel(i.e. Patentable) product
and testing ideas
*	This Engineer will expand the relationships between NSP and
other HP organizations
*	This Engineer will represent ISS and NSP at industry events


Education/Experience: 
*	Must have CCNA,  Prefer CCIE
*	Must have 8 years of professional Network and/or System
Administration experience
*	Bachelors degree and 8 years of professional experience or a
Masters with 5 years of professional experience is preferred

Demonstrable Knowledge:  
*	Windows Networking
*	Require at least 4 of the following
o	Linux/UNIX Networking
o	NetWare Networking
o	Infiniband
o	iSCSI
o	SNMP
o	Creating and Managing Complex Network Topologies
o	Application-level Programming(BASH, C/C++, VC++, Java, etc.)
o	IXIA or SmartBits
o	Sniffers

Contact Ted Wolf
ted.a.wolf@hp.com

From gil@kloepfer.org Sat Nov 12 08:18:36 2005
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From: Gil Kloepfer <aaug@kloepfer.org>
To: austin-asterisk-users-group@bybent.com
Subject: Re: [A*UG] Looking for a Sr. Networking Engineer
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On Thu, Nov 03, 2005 at 04:52:00PM -0600, Wolf, Ted (ISS/Austin TX) wrote:
> HP's Network System Products organization is seeking a networking expert
> for our Product Assurance Team.
[...]

If this is the kind of stuff that is going to start getting posted
to this list, then I will be getting off of it pretty soon.

I get enough e-mail without job offers I don't want and never asked for.

---
Gil Kloepfer
aaug@kloepfer.org
From fmouse-aaug@fmp.com Sat Nov 12 10:44:35 2005
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Whoever is the list admin for this list should direct people who want to
post job offers to one of to other lists rather than this one.

jobs@lists.cactus.org
ctlug@ctlug.org

Both are appropriate forums for this.  I agree with Gil, that the A*UG list
should be well focused on its topic and that job offers here are quite OT.

Thus spake Gil Kloepfer on Sat, Nov 12, 2005 at 08:18:32AM CST
> On Thu, Nov 03, 2005 at 04:52:00PM -0600, Wolf, Ted (ISS/Austin TX) wrote:
> > HP's Network System Products organization is seeking a networking expert
> > for our Product Assurance Team.
> [...]
> 
> If this is the kind of stuff that is going to start getting posted
> to this list, then I will be getting off of it pretty soon.
> 
> I get enough e-mail without job offers I don't want and never asked for.

-- 
Lindsay Haisley       | "Everything works    |     PGP public key
FMP Computer Services |       if you let it" |      available at
512-259-1190          |    (The Roadie)      | <http://www.fmp.com/pubkeys>
http://www.fmp.com    |                      |
From wwalker@behemoth.gnumber.com Mon Nov 21 15:07:50 2005
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Hi All,

It's Astrerisk meeting night.  However, our resident expert, Chris
Tooley, probably will not be able to attend this evening.  I'll be here
to let people in and stare glassy eyed at you pretending I know nothing
(not a far stretch :).  I can answer some questions, especially
programming, but the FXS/FXO and "which phone" questions are out of my
knowledge.  Hopefully someone else will show up who can field these type
of questions.

So, who all's coming?  If you tell me ahead of time, I'll be more likely
to notice someone's missing and come let them in.

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net
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Wayne Walker wrote:
> Hi All,
> 
> It's Astrerisk meeting night.  However, our resident expert, Chris
> Tooley, probably will not be able to attend this evening.  I'll be here
> to let people in and stare glassy eyed at you pretending I know nothing
> (not a far stretch :).  I can answer some questions, especially
> programming, but the FXS/FXO and "which phone" questions are out of my
> knowledge.  Hopefully someone else will show up who can field these type
> of questions.
> 
> So, who all's coming?  If you tell me ahead of time, I'll be more likely
> to notice someone's missing and come let them in.
> 
  Me.

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Subject: Re: [A*UG] Meeting Tonight
From: Les Elliott <lelliott5@gmail.com>
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I have to miss tonight (was really looking forward to it).  Family
matters.  Hope everyone has a great Thanksgiving.
Les


On Mon, 2005-11-21 at 15:07 -0600, Wayne Walker wrote:
> Hi All,
> 
> It's Astrerisk meeting night.  However, our resident expert, Chris
> Tooley, probably will not be able to attend this evening.  I'll be here
> to let people in and stare glassy eyed at you pretending I know nothing
> (not a far stretch :).  I can answer some questions, especially
> programming, but the FXS/FXO and "which phone" questions are out of my
> knowledge.  Hopefully someone else will show up who can field these type
> of questions.
> 
> So, who all's coming?  If you tell me ahead of time, I'll be more likely
> to notice someone's missing and come let them in.
> 

From jimfreeze@gmail.com Mon Nov 21 17:45:23 2005
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On 11/21/05, Wayne Walker <wwalker@bybent.com> wrote:

> It's Astrerisk meeting night.  However, our resident expert, Chris

Where?

--
Jim Freeze

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At my office in the MCC building.  Check out 

http://aaug.bybent.com/

On Mon, Nov 21, 2005 at 05:39:08PM -0600, Jim Freeze wrote:
> On 11/21/05, Wayne Walker <wwalker@bybent.com> wrote:
> 
> > It's Astrerisk meeting night.  However, our resident expert, Chris
> 
> Where?
> 
> --
> Jim Freeze

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
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Subject: [A*UG] Would DECT cordless phones work with Asterisk and VOIP?
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I've just heard about DECT which is used for about 50 million phones in 
Europe and is just starting to appear in the US.

DECT stands for Digitally Enhanced Cordless Telephone
and supposedly has much greater range than other cordless telephony.
Additionally, you can purchase repeaters that will greatly range.

Since reading about poor audio quality and echo issues caused by 
repeated conversions as the signal traverses the path from the (possibly 
POTS analogue) station, over tcp/ip and to destination,
would DECT (another digital form) agravate this?

In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only 
work over about 35 feet indoors - not enough for a large house.

Does anyone have any hands-on experience with DECT?

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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I am on the front edge of what, to me, feels like a large asterisk =
roll-out
(20 lines + 60 extensions), and since this is the first month I've been =
in
town during the week of the users group meeting was hoping to attend =
this
month. Is there likely to be a January meeting, or has this user group =
met a
quiet end?

=20

Jim Kuligowski

jimkski@jimkski.net


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am on the front edge of what, to me, feels like a =
large
asterisk roll-out (20 lines + 60 extensions), and since this is the =
first month
I&#8217;ve been in town during the week of the users group meeting was =
hoping
to attend this month. Is there likely to be a January meeting, or has =
this user
group met a quiet end?</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Jim Kuligowski</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>jimkski@jimkski.net</span></font></p>

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From: Wayne Walker <wwalker@bybent.com>
To: jimkski <jimkski@trinitypublications.com>
Subject: Re: [A*UG] No December meeting?
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There waas a November meeting though it was only a couple of folks.  I
plan for there to be a January meeting, just haven't planned it yet.

On Wed, Dec 21, 2005 at 02:02:10PM -0600, jimkski wrote:
> 
>    I am on the front edge of what, to me, feels like a large asterisk roll-out
>    (20 lines + 60 extensions), and since this is the first month Iâ€™ve been in
>    town during the week of the users group meeting was hoping to attend this
>    month. Is there likely to be a January meeting, or has this user group met a
>    quiet end?
> 
> 
>    Jim Kuligowski
> 
>    jimkski@jimkski.net

> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group


-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
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From carey@itfreedom.com Thu Dec 22 11:32:42 2005
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From: "Carey Jung" <carey@itfreedom.com>
To: "'Wayne Walker'" <wwalker@bybent.com>,
   "'jimkski'" <jimkski@trinitypublications.com>
Subject: RE: [A*UG] No December meeting?
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I'd be happy to chat about our experiences and lessons learned so far.  We have two customers and about a hundred users on board, with another 300 users coming on board in the next few weeks.   We've learned a ton.   When is the next meeting?

> -----Original Message-----
> From: austin-asterisk-users-group-bounces@bybent.com 
> [mailto:austin-asterisk-users-group-bounces@bybent.com] On 
> Behalf Of Wayne Walker
> Sent: Thursday, December 22, 2005 11:19 AM
> To: jimkski
> Cc: austin-asterisk-users-group@bybent.com
> Subject: Re: [A*UG] No December meeting?
> 
> There waas a November meeting though it was only a couple of 
> folks.  I plan for there to be a January meeting, just 
> haven't planned it yet.
> 
> On Wed, Dec 21, 2005 at 02:02:10PM -0600, jimkski wrote:
> > 
> >    I am on the front edge of what, to me, feels like a 
> large asterisk roll-out
> >    (20 lines + 60 extensions), and since this is the first 
> month IÃ¢â‚¬â„¢ve been in
> >    town during the week of the users group meeting was 
> hoping to attend this
> >    month. Is there likely to be a January meeting, or has 
> this user group met a
> >    quiet end?
> > 
> > 
> >    Jim Kuligowski
> > 
> >    jimkski@jimkski.net
> 
> > _______________________________________________
> > Austin-Asterisk-Users-Group mailing list 
> > Austin-Asterisk-Users-Group@bybent.com
> > http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
> 
> -- 
> 
> Wayne Walker
> 
> www.unwiredbuyer.com - when you just can't be by the computer
> 
> wwalker@bybent.com                    Do you use Linux?!
> http://www.bybent.com                 Get Counted!  
> http://counter.li.org/
> Perl - http://www.perl.org/           Perl User Groups - 
> http://www.pm.org/
> Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
> IRC:     wwalker on freenode.net
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list 
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 


From Greg@towerz.com Thu Dec 22 12:25:01 2005
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I will surely show up for the next meeting. I have many questions and a big rollout as well.

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225
 

-----Original Message-----
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Subject: Austin-Asterisk-Users-Group Digest, Vol 3, Issue 1

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Today's Topics:

   1. No December meeting? (jimkski)
   2. Re: No December meeting? (Wayne Walker)
   3. RE: No December meeting? (Carey Jung)


----------------------------------------------------------------------

Message: 1
Date: Wed, 21 Dec 2005 14:02:10 -0600
From: "jimkski" <jimkski@trinitypublications.com>
Subject: [A*UG] No December meeting?
To: <austin-asterisk-users-group@bybent.com>
Message-ID: <001701c60669$6e2f3c80$0c96a8c0@jimxp>
Content-Type: text/plain; charset="us-ascii"

I am on the front edge of what, to me, feels like a large asterisk roll-out
(20 lines + 60 extensions), and since this is the first month I've been in
town during the week of the users group meeting was hoping to attend this
month. Is there likely to be a January meeting, or has this user group met a
quiet end?

 

Jim Kuligowski

jimkski@jimkski.net

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------------------------------

Message: 2
Date: Thu, 22 Dec 2005 11:18:51 -0600
From: Wayne Walker <wwalker@bybent.com>
Subject: Re: [A*UG] No December meeting?
To: jimkski <jimkski@trinitypublications.com>
Cc: austin-asterisk-users-group@bybent.com
Message-ID: <20051222171851.GA4386@bybent.com>
Content-Type: text/plain; charset=unknown-8bit

There waas a November meeting though it was only a couple of folks.  I
plan for there to be a January meeting, just haven't planned it yet.

On Wed, Dec 21, 2005 at 02:02:10PM -0600, jimkski wrote:
> 
>    I am on the front edge of what, to me, feels like a large asterisk roll-out
>    (20 lines + 60 extensions), and since this is the first month IÃ¢â‚¬â„¢ve been in
>    town during the week of the users group meeting was hoping to attend this
>    month. Is there likely to be a January meeting, or has this user group met a
>    quiet end?
> 
> 
>    Jim Kuligowski
> 
>    jimkski@jimkski.net

> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group


-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net

------------------------------

Message: 3
Date: Thu, 22 Dec 2005 11:27:35 -0600
From: "Carey Jung" <carey@itfreedom.com>
Subject: RE: [A*UG] No December meeting?
To: "'Wayne Walker'" <wwalker@bybent.com>,   "'jimkski'"
	<jimkski@trinitypublications.com>
Cc: austin-asterisk-users-group@bybent.com
Message-ID: <00f101c6071c$ffca6210$b816010a@carey>
Content-Type: text/plain;	charset="UTF-8"

I'd be happy to chat about our experiences and lessons learned so far.  We have two customers and about a hundred users on board, with another 300 users coming on board in the next few weeks.   We've learned a ton.   When is the next meeting?

> -----Original Message-----
> From: austin-asterisk-users-group-bounces@bybent.com 
> [mailto:austin-asterisk-users-group-bounces@bybent.com] On 
> Behalf Of Wayne Walker
> Sent: Thursday, December 22, 2005 11:19 AM
> To: jimkski
> Cc: austin-asterisk-users-group@bybent.com
> Subject: Re: [A*UG] No December meeting?
> 
> There waas a November meeting though it was only a couple of 
> folks.  I plan for there to be a January meeting, just 
> haven't planned it yet.
> 
> On Wed, Dec 21, 2005 at 02:02:10PM -0600, jimkski wrote:
> > 
> >    I am on the front edge of what, to me, feels like a 
> large asterisk roll-out
> >    (20 lines + 60 extensions), and since this is the first 
> month IÃƒÂ¢Ã¢â€šÂ¬Ã¢â€žÂ¢ve been in
> >    town during the week of the users group meeting was 
> hoping to attend this
> >    month. Is there likely to be a January meeting, or has 
> this user group met a
> >    quiet end?
> > 
> > 
> >    Jim Kuligowski
> > 
> >    jimkski@jimkski.net
> 
> > _______________________________________________
> > Austin-Asterisk-Users-Group mailing list 
> > Austin-Asterisk-Users-Group@bybent.com
> > http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
> 
> -- 
> 
> Wayne Walker
> 
> www.unwiredbuyer.com - when you just can't be by the computer
> 
> wwalker@bybent.com                    Do you use Linux?!
> http://www.bybent.com                 Get Counted!  
> http://counter.li.org/
> Perl - http://www.perl.org/           Perl User Groups - 
> http://www.pm.org/
> Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
> IRC:     wwalker on freenode.net
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list 
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 



------------------------------

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End of Austin-Asterisk-Users-Group Digest, Vol 3, Issue 1
*********************************************************


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On 12/22/05, Greg Shapiro <Greg@towerz.com> wrote:
> I will surely show up for the next meeting. I have many questions and a big rollout as well.

I will go to a December meeting.
If it is postponed till Jan, would be nice to have it early.

--
Jim Freeze

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Anyone running Asterisk on OpenWrt?

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Joe Barr wrote:
> Anyone running Asterisk on OpenWrt?
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
I'd _like_ to Joe and have a nice WRT version 4 but I've read the 
openWRT site and don't really understand how to get started.

Guess I need a WRT guru.

In the meantime, I have Asterisk@Home running on a desktop tower and am 
waiting for my Grandstream GXP2000 phones to arrive to get started with 
that.

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Subject: Re: [A*UG] Asterisk on OpenWrt
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On Wed, 2005-12-28 at 21:49 -0600, Larry Alkoff wrote:

>I'd _like_ to Joe and have a nice WRT version 4 but I've read the 
>openWRT site and don't really understand how to get started.

>Guess I need a WRT guru.

>In the meantime, I have Asterisk@Home running on a desktop tower and
>am waiting for my Grandstream GXP2000 phones to arrive to get started
>with that.

So far, so good.  I have the WRT54GS in operation now with RC4 of
OpenWrt White Russian, and using the new GUI admin console, installed
Asterisk in about 1 minute from the OpenWrt repository.

I guess this means I need to get a SIP phone and learn how to use
Asterisk.

Joe


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If someone brings one, we should be able to do the whole install in
about 30 minutes.

On Wed, Dec 28, 2005 at 09:49:30PM -0600, Larry Alkoff wrote:
> Joe Barr wrote:
> >Anyone running Asterisk on OpenWrt?
> >
> >_______________________________________________
> >Austin-Asterisk-Users-Group mailing list
> >Austin-Asterisk-Users-Group@bybent.com
> >http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> >
> I'd _like_ to Joe and have a nice WRT version 4 but I've read the 
> openWRT site and don't really understand how to get started.
> 
> Guess I need a WRT guru.
> 
> In the meantime, I have Asterisk@Home running on a desktop tower and am 
> waiting for my Grandstream GXP2000 phones to arrive to get started with 
> that.
> 
> -- 
> Larry Alkoff N2LA - Austin TX
> Using Thunderbird on Slackware Linux
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
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Two Questions:

1) When is the next meeting?
2) Can someone suggest a URL to purchase a good phone?

Thanks


--
Jim Freeze

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Two Questions:<br><br>1) When is the next meeting?<br>2) Can someone sugges=
t a URL to purchase a good phone?<br><br>Thanks<br><br><br>-- <br>Jim Freez=
e

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Wow Wayne - 30 minutes! I had no idea it could be so easy.
So when could I bring you my WRT54G version 4?

What else would I need?  My only (at the moment) Grandstream GXP-2000? 
Would you install Asterisk or Asterisk@home?  Passwords etc for my ISP 
Telesip?

Larry



Wayne Walker wrote:
> If someone brings one, we should be able to do the whole install in
> about 30 minutes.
> 
> On Wed, Dec 28, 2005 at 09:49:30PM -0600, Larry Alkoff wrote:
> 
>>Joe Barr wrote:
>>
>>>Anyone running Asterisk on OpenWrt?
>>>
>>>_______________________________________________
>>>Austin-Asterisk-Users-Group mailing list
>>>Austin-Asterisk-Users-Group@bybent.com
>>>http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
>>>
>>
>>I'd _like_ to Joe and have a nice WRT version 4 but I've read the 
>>openWRT site and don't really understand how to get started.
>>
>>Guess I need a WRT guru.
>>
>>In the meantime, I have Asterisk@Home running on a desktop tower and am 
>>waiting for my Grandstream GXP2000 phones to arrive to get started with 
>>that.
>>
>>-- 
>>Larry Alkoff N2LA - Austin TX
>>Using Thunderbird on Slackware Linux
>>_______________________________________________
>>Austin-Asterisk-Users-Group mailing list
>>Austin-Asterisk-Users-Group@bybent.com
>>http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> 
> 


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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I received my BT-101 today and hope to hack out a working config for
Asterisk over the weekend.  Any suggestions on a provider to sign up
with?

Thanks,
Joe

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Hi Guys. I am about to set up a asterisk system for a small office and
am wondering when the next meeting is. I have a lot of questions before
I go and spend a lot of money no phones and equipment.

If anyone is interested, I can give a post mortem talk when it is all  
done.
BTW, I plan to script Asterisk with Ragi.

Jim

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Joe Barr wrote:
> I received my BT-101 today and hope to hack out a working config for
> Asterisk over the weekend.  Any suggestions on a provider to sign up
> with?
> 
> Thanks,
> Joe


Joe I'm using Telasip at
htty://www.telesip.com

with forums at
http://www.telasipforums.com/index.php

They are inexpensive and feature excellent support.
The support guy likes to talk to customers and resolve problems and
there is no wait.
About $15. monthly for unlimited 24/7 US (and Can?) service 2 lines.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From rabel@randy.abel.name Sat Jan  7 13:00:21 2006
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I am also preparing to setup my own Asterix install and am interested in 
any user group meetings that may be forthcoming.  I'm happy to prepare 
install info and post mortem once I get it working.

I have a very small business that is just getting started.  I purchase 
many different products from China and would like to be able to 
make/receive international calls using cheap VoIP service.  I sell my 
products nationwide so an inexpensive nationwide toll-free number is 
also a goal.  Additional interesting possibilities include custom phone 
numbers for marketing campaigns so I can track marketing effectiveness 
easily, local phone numbers in major markets and (since I'm dreaming 
here), perhaps international numbers when I expand to international 
sales.  Fax support, voicemail, call routing, IP phone for when I'm 
travelling, etc.

Basically, a complete phone system for my little business that also 
would allow me to do personal stuff such as a seperate phone number for 
my daughter, voicemail and fax accessibility over the web, message 
notification to my cell phone, etc.

Randy

From jimfreeze@gmail.com Sat Jan  7 13:17:00 2006
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In case the list hasn't seen the video by Systm, here is the info.
I found it very interesting.

On Jan 5, 2006, at 10:30 PM, Steven Rogers wrote:

> Kevin Rose's Systm video blog has an episode today about Asterisk.
> Its an open source PBX, but its much more interesting than it sounds.
>
> http://revision3.com/systm/asterisk/


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; =
-khtml-line-break: after-white-space; ">In case the list hasn't seen the =
video by Systm, here is the info.<DIV>I found it very =
interesting.</DIV><DIV><BR><DIV><DIV>On Jan 5, 2006, at 10:30 PM, Steven =
Rogers wrote:</DIV><BR class=3D"Apple-interchange-newline"><BLOCKQUOTE =
type=3D"cite"><P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT =
face=3D"Helvetica" size=3D"5" style=3D"font: 18.0px Helvetica">Kevin =
Rose's Systm video blog has an episode today about Asterisk. <SPAN =
class=3D"Apple-converted-space">=A0</SPAN></FONT></P> <P style=3D"margin: =
0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"5" =
style=3D"font: 18.0px Helvetica">Its an open source PBX, but its much =
more interesting than it sounds.</FONT></P> <P style=3D"margin: 0.0px =
0.0px 0.0px 0.0px; font: 18.0px Helvetica; min-height: 22.0px"><BR></P> =
<P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" =
size=3D"5" style=3D"font: 18.0px Helvetica"><A =
href=3D"http://revision3.com/systm/asterisk/">http://revision3.com/systm/a=
sterisk/</A></FONT></P> </BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>=

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I believe the official meeting time is the third Monday of the month.
Last month was horrible timing for me.  However, I think this month we
can get back to the program.  That puts the meeting date on the 16th.
We met the first time at 7PM, is there a huge problem with doing 6:30?

I'll suggest a meeting comparing SIP endpoint devices.  I can bring in
an ATA device from 8x8, a SIP phone by Snom and we have several Polycom
IP500's at the office.  Anyone got anything else they can bring in?


On Sat, 2006-01-07 at 12:52 -0600, Randy Abel wrote:
> I am also preparing to setup my own Asterix install and am interested in 
> any user group meetings that may be forthcoming.  I'm happy to prepare 
> install info and post mortem once I get it working.
> 
> I have a very small business that is just getting started.  I purchase 
> many different products from China and would like to be able to 
> make/receive international calls using cheap VoIP service.  I sell my 
> products nationwide so an inexpensive nationwide toll-free number is 
> also a goal.  Additional interesting possibilities include custom phone 
> numbers for marketing campaigns so I can track marketing effectiveness 
> easily, local phone numbers in major markets and (since I'm dreaming 
> here), perhaps international numbers when I expand to international 
> sales.  Fax support, voicemail, call routing, IP phone for when I'm 
> travelling, etc.
> 
> Basically, a complete phone system for my little business that also 
> would allow me to do personal stuff such as a seperate phone number for 
> my daughter, voicemail and fax accessibility over the web, message 
> notification to my cell phone, etc.
> 
> Randy
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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It only took me a day or two, but I seem to have borked my BT-101.  I no
longer hear a dial tone when I lift the handset or press speaker.
Anyone have a magic tip for me on how to get it back?  Software reset to
defaults doesn't seem to have cured it.

From gmahan@dmibs.com Mon Jan  9 17:46:56 2006
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	Need to purchase Sipura ATA 2100 or 3000 ASAP -- ?? on Local Vendor
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* Group,
    Does anyone know where one can purchase ATAs (Sipura 2100/3000) and IP
Phones (Polycom/Linksys+Sipura) here in Austin? Is there similar to
voipsupply.com locally? 
 
    Additionally, does anyone have a used Sipura 2000/2100/3000 ATA they
would like to sell? I need to purchase one ASAP.
 
Thanks
 
George
512-924-1243 

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<BODY>
<DIV dir=3Dltr align=3Dleft>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial size=3D2>*=20
Group,</FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial =
size=3D2>&nbsp;&nbsp;&nbsp;=20
Does anyone know where one can purchase ATAs (Sipura 2100/3000) and IP =
Phones=20
(Polycom/Linksys+Sipura) here in Austin? Is there similar to =
voipsupply.com=20
locally? </FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D250090723-09012006>&nbsp;&nbsp;&nbsp; <FONT =
face=3DArial=20
size=3D2>Additionally, does anyone have a used Sipura 2000/2100/3000 ATA =
they=20
would like to sell? I need to purchase one ASAP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
size=3D2>Thanks</FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
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size=3D2>George</FONT></SPAN></DIV>
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From joe@pjprimer.com Mon Jan  9 22:58:07 2006
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W00t!

I've gotten far enough to be able to receive calls from NuFone via
Asterisk on the router and ring them thru to the BT-101.  Hooray!

Next hurdle is getting the configuration right so that I can dial out
thru NuFone.

My problem with the dial tone was solved by _correctly_ resetting the
phone to factory defaults.  For over a day I thought I was resetting the
phone, when in fact I never did.

From gemahan@mac.com Mon Jan  9 17:28:29 2006
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* Group,
    Does anyone know where one can purchase ATAs (Sipura 2100/3000) and IP
Phones (Polycom/Linksys+Sipura) here in Austin? Is there similar to
voipsupply.com locally? 
 
    Additionally, does anyone have a used Sipura 2000/2100/3000 ATA they
would like to sell? I need to purchase one ASAP.
 
Thanks
 
George
512-924-1243 
 

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<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial size=3D2>*=20
Group,</FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial =
size=3D2>&nbsp;&nbsp;&nbsp;=20
Does anyone know where one can purchase ATAs (Sipura 2100/3000) and IP =
Phones=20
(Polycom/Linksys+Sipura) here in Austin? Is there similar to =
voipsupply.com=20
locally? </FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D250090723-09012006>&nbsp;&nbsp;&nbsp; <FONT =
face=3DArial=20
size=3D2>Additionally, does anyone have a used Sipura 2000/2100/3000 ATA =
they=20
would like to sell? I need to purchase one ASAP.</FONT></SPAN></DIV>
<DIV><SPAN class=3D250090723-09012006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
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From dbii@interaction.net Tue Jan 10 14:09:57 2006
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Subject: [A*UG] Meeting on Monday due to Holiday?
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Wayne and Chris-

Are you still planning on a meeting on Monday for January since it is a 
holiday?

If so, what topic?

David
-----------------------------------------------------------------------
David H. Bluestein II
President & Lead Developer         dbii@interaction.net
ii, inc.                                                  
http://www.interaction.net
     -- Specializing in Interactive, Database Driven Websites --

From wwalker@behemoth.gnumber.com Tue Jan 10 14:48:37 2006
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First off, the next meeting is scheduled for the gNumber office, on
Monday the 16th of January.  at 6:30.

I know Chris already stated this, but I know we've some new members on
the list and that the website isn't up to date (my next task).

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net
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On Tue, 2006-01-10 at 13:18 -0600, David Bluestein II wrote:
> Wayne and Chris-
> 
> Are you still planning on a meeting on Monday for January since it is a 
> holiday?
> 
It is?

> If so, what topic?
> 
SIP hardware and devices

> David
> -----------------------------------------------------------------------
> David H. Bluestein II
> President & Lead Developer         dbii@interaction.net
> ii, inc.                                                  
> http://www.interaction.net
>      -- Specializing in Interactive, Database Driven Websites --
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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Joe Barr wrote:
> W00t!
> 
> I've gotten far enough to be able to receive calls from NuFone via
> Asterisk on the router and ring them thru to the BT-101.  Hooray!
> 
> Next hurdle is getting the configuration right so that I can dial out
> thru NuFone.
> 
> My problem with the dial tone was solved by _correctly_ resetting the
> phone to factory defaults.  For over a day I thought I was resetting the
> phone, when in fact I never did.
> 

Joe I'm also working on Asterisk setup using Asterisk@Home version 2.2.
If you are using that or 2.1 there is an very excellent tutorial about 
setting it up at http://mundy.org/blog/index.php?p=93

It goes into great about setting up AAH.

I'm in the stage of just having setup a computer for AAH, got it loaded 
and am about to start configuring it according to the article.

If you want to talk to me about Asterisk or anything else
give me a call at home.  I'm in the book and am there during the day now.

What is your call?

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From n5xwb@hal-pc.org Tue Jan 10 17:39:11 2006
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FYI I ran across this site

"http://www.voip-info.org/wiki-Asterisk+phones"


Larry Alkoff wrote:
> Joe Barr wrote:
> 
>>W00t!
>>
>>I've gotten far enough to be able to receive calls from NuFone via
>>Asterisk on the router and ring them thru to the BT-101.  Hooray!
>>
>>Next hurdle is getting the configuration right so that I can dial out
>>thru NuFone.
>>
>>My problem with the dial tone was solved by _correctly_ resetting the
>>phone to factory defaults.  For over a day I thought I was resetting the
>>phone, when in fact I never did.
>>
> 
> 
> Joe I'm also working on Asterisk setup using Asterisk@Home version 2.2.
> If you are using that or 2.1 there is an very excellent tutorial about 
> setting it up at http://mundy.org/blog/index.php?p=93
> 
> It goes into great about setting up AAH.
> 
> I'm in the stage of just having setup a computer for AAH, got it loaded 
> and am about to start configuring it according to the article.
> 
> If you want to talk to me about Asterisk or anything else
> give me a call at home.  I'm in the book and am there during the day now.
> 
> What is your call?
> 
> Larry
> 


-- 
73 de Donn Washburn  Hpage:" http://www.hal-pc.org/~n5xwb "
Ham Callsign N5XWB   Email: " n5xwb@hal-pc.org "
307 Savoy St.        HAMs : " n5xwb@arrl.net "
Sugar Land, TX 77478 BMW MOA #: 4146 - Ambassador
LL# 1.281.242.3256   VoIP via Skype: n5xwbg
       " http://counter.li.org " #279316
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From: Jim Freeze <jimfreeze@gmail.com>
Subject: Re: [A*UG] Meeting on Monday due to Holiday?
Date: Tue, 10 Jan 2006 18:31:01 -0600
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Can someone post directions to the meeting?

Thanks

Jim

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Donn Washburn wrote:
> FYI I ran across this site
> 
> "http://www.voip-info.org/wiki-Asterisk+phones"

Hi Donn.

Yes, voip-info.org is another great site.

The reason I was so pleased with the mundy site is that
this particular article was specific to Asterisk@Home 2.2 which
is the latest version of AAH, which is what I'm setting up.
His setup info is very detailed on 2.2 and other aspects of Asterisk
and what you can do with it.

He also has several excellent urls where Ward Mundy gives Asterisk info
all under the name of 'Nerd Vittles'.  It's well worth looking at his 
information.

73, Larry



> 
> Larry Alkoff wrote:
> 
>> Joe Barr wrote:
>>
>>> W00t!
>>>
>>> I've gotten far enough to be able to receive calls from NuFone via
>>> Asterisk on the router and ring them thru to the BT-101.  Hooray!
>>>
>>> Next hurdle is getting the configuration right so that I can dial out
>>> thru NuFone.
>>>
>>> My problem with the dial tone was solved by _correctly_ resetting the
>>> phone to factory defaults.  For over a day I thought I was resetting the
>>> phone, when in fact I never did.
>>>
>>
>>
>> Joe I'm also working on Asterisk setup using Asterisk@Home version 2.2.
>> If you are using that or 2.1 there is an very excellent tutorial about 
>> setting it up at http://mundy.org/blog/index.php?p=93
>>
>> It goes into great about setting up AAH.
>>
>> I'm in the stage of just having setup a computer for AAH, got it 
>> loaded and am about to start configuring it according to the article.
>>
>> If you want to talk to me about Asterisk or anything else
>> give me a call at home.  I'm in the book and am there during the day now.
>>
>> What is your call?
>>
>> Larry
>>
> 
> 


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Hi all,

Just a reminder that we're scheduled to meet Monday January 16th.

That's tomorrow or today according to whether you read this on Sunday or
not...

Seems thtat quite a few folks will be bringing voip phones with them

I think one person was going to bring a Linksys WRT54G to install on
also.

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
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On 1/15/06 9:34 AM, "Wayne Walker" <wwalker@bybent.com> wrote:

> Seems thtat quite a few folks will be bringing voip phones with them

I plan on bringing my Sipura SPA-3000 [1] and Dock-n-Talk Cell Phone Station
[2].  Hopefully, someone can help me configure the SPA-3000 to allow calls
coming in on the PSTN line (using the dock-n-talk) to make calls out through
the voip line (in this case: Broadvoice's BYOD [3] plan).

If I can get this working, I'll be able to use the unlimited t-mobile to
t-mobile plan to effectively make free calls to Canada from my cell phone.
Woot!

[1] http://store.voxilla.com/customer/product.php?productid=16144
[2] http://store.voxilla.com/customer/product.php?productid=16136
[3] http://www.broadvoice.com/rateplans_byod.html


From Greg@towerz.com Sun Jan 15 14:54:46 2006
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Could I get directions to the meeting.

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225
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Today's Topics:

   1. Meeting Monday, January 16th (Wayne Walker)


----------------------------------------------------------------------

Message: 1
Date: Sun, 15 Jan 2006 09:34:35 -0600
From: Wayne Walker <wwalker@bybent.com>
Subject: [A*UG] Meeting Monday, January 16th
To: austin-asterisk-users-group@bybent.com
Message-ID: <20060115153435.GG13173@bybent.com>
Content-Type: text/plain; charset=us-ascii

Hi all,

Just a reminder that we're scheduled to meet Monday January 16th.

That's tomorrow or today according to whether you read this on Sunday or
not...

Seems thtat quite a few folks will be bringing voip phones with them

I think one person was going to bring a Linksys WRT54G to install on
also.

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!
http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups -
http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net

------------------------------

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End of Austin-Asterisk-Users-Group Digest, Vol 4, Issue 6
*********************************************************


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Address of the meeting?

Mike Gunter
VoxFlow



-----Original Message-----
From: austin-asterisk-users-group-bounces@bybent.com
[mailto:austin-asterisk-users-group-bounces@bybent.com]
On Behalf Of Greg Shapiro
Sent: Sunday, January 15, 2006 2:52 PM
To: austin-asterisk-users-group@bybent.com
Subject: [A*UG] RE: Austin-Asterisk-Users-Group Digest,
Vol 4, Issue 6

Could I get directions to the meeting.

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225
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Today's Topics:

   1. Meeting Monday, January 16th (Wayne Walker)


-------------------------------------------------------
---------------

Message: 1
Date: Sun, 15 Jan 2006 09:34:35 -0600
From: Wayne Walker <wwalker@bybent.com>
Subject: [A*UG] Meeting Monday, January 16th
To: austin-asterisk-users-group@bybent.com
Message-ID: <20060115153435.GG13173@bybent.com>
Content-Type: text/plain; charset=us-ascii

Hi all,

Just a reminder that we're scheduled to meet Monday
January 16th.

That's tomorrow or today according to whether you read
this on Sunday or
not...

Seems thtat quite a few folks will be bringing voip
phones with them

I think one person was going to bring a Linksys WRT54G
to install on
also.

--

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the
computer

wwalker@bybent.com                    Do you use
Linux?!
http://www.bybent.com                 Get Counted!
http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups
-
http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:
lwwalkerbybent
IRC:     wwalker on freenode.net

------------------------------

_______________________________________________
Austin-Asterisk-Users-Group mailing list
Austin-Asterisk-Users-Group@bybent.com
http://buzzard.onr.com/mailman/listinfo/austin-asterisk
-users-group
AAUG Web Site: http://aaug.bybent.com/

End of Austin-Asterisk-Users-Group Digest, Vol 4, Issue
6
*******************************************************
**


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From wwalker@behemoth.gnumber.com Mon Jan 16 12:24:59 2006
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The meeting is tonight at 6:30 )PM at gNumber's office.

Please go to http://aaug.bybent.com/ for the latest info and
directions.

NOTE: WARNING: You MUST call me to come get you.  (512-669-2456)

NOTE: If you get out of your car and you can't look across the street
and see Burger King, you are on the wrong side of the building.

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!  http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups - http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
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From emaillists@simondorfman.com Mon Jan 16 23:42:56 2006
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At the meeting tonight, I mentioned an eFax alternative, but couldn't
remember the name and promised to look it up and email it to the list.  Here
it is:
http://maxemail.com/fax/services_features.html

I'm using the $15/year "MaxEmail® Lite" which has been working fine for the
5 months or so that I've been using them.

Simon



From lenny@VoIPing.com Tue Jan 17 00:12:14 2006
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Subject: [A*UG] Good meeting everyone
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It was good getting together tonight and it's great to see the interest
in Asterisk.  Below is my contact info for those who asked.  I'll be
happy to participate in future talks discussions, tutorials, etc.  

In an aside, for Chris and Wayne, I signed up for UnwiredBuyer (using 
"asterisk" as the promo code).  Cool service!  Just got my first call 
from the service on something in my watch list...   didn't bid, but 
will in the future!!  Hope you guys make it big someday with those 
millions for a startup.  Working for startups myself, some who've 
failed and some who haven't... I know what it's like to pour everything 
into a company!

Some interest to the group, here's some links for ya'll.

This might be a little dated (back from 2003, but most of the stuff 
is fairly relevant).  My CACTUS presentation on VoIP, etc.  
	http://www.voiping.com/cactus/

This talk is from Astricon 2004.  "Bye Avaya, Hello Asterisk..."
	http://www.astricon.net/conf2004/docs/3-lenny-tropiano.pdf

Photos from Astricon 2004.
	http://photos.tropiano.org/gallery/astricon-2004

Lenny w/ Marc Spencer ...
(Gosh, I'm glad I lost some weight!!) :) :)
	http://photos.tropiano.org/gallery/astricon-2004-1/IMG_0008

With the imfamous, "Allison, voice of Asterisk"
	http://photos.tropiano.org/gallery/astricon-2004-1/IMG_0029

CACTUS is this Thursday, 
	http://www.cactus.org/CACTUS/Newsletter/cn200601.html

Happy VoIP'ing,

L.
-- 
Lenny Tropiano                          E-mail: lenny@voiping.com
Partner, Networking Specialist          Pager:  pager-lenny@voiping.com
VoIPing, LLC                            URL:    http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-0867   Mobile: 512-698-VOIP [8647] 
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On Tue, Jan 17, 2006 at 12:06:33AM -0600, Lenny Tropiano wrote:
> Some interest to the group, here's some links for ya'll.

Not to be entirely outdone by Lenny, here's a link to my Asterisk
presentation at CACTUS as well.  However, dated as it is, it
does have a working set of config files in a simple arrangement
that may help to get some folks started.

        http://www.kloepfer.org/gc2/cactus.html

(the Asterisk presentation is #2).

Be sure to look at the presentation notes before the config files
because the conceptual configuration is described in there.  Again,
these are dated, and I can think of better ways of doing some of
the things in the dialplan now.

It was definitely good meeting all of you, and I felt that walking
away just knowing that the newer Polycom IP500 sets could do PoE
made the meeting worthwhile (although there was so much more discussed).

---
Gil Kloepfer
aaug@kloepfer.org
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Second that.
Meeting Address?



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[mailto:austin-asterisk-users-group-bounces@bybent.com]On Behalf Of Greg
Shapiro
Sent: Sunday, January 15, 2006 2:52 PM
To: austin-asterisk-users-group@bybent.com
Subject: [A*UG] RE: Austin-Asterisk-Users-Group Digest, Vol 4, Issue 6


Could I get directions to the meeting.

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225
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Today's Topics:

   1. Meeting Monday, January 16th (Wayne Walker)


----------------------------------------------------------------------

Message: 1
Date: Sun, 15 Jan 2006 09:34:35 -0600
From: Wayne Walker <wwalker@bybent.com>
Subject: [A*UG] Meeting Monday, January 16th
To: austin-asterisk-users-group@bybent.com
Message-ID: <20060115153435.GG13173@bybent.com>
Content-Type: text/plain; charset=us-ascii

Hi all,

Just a reminder that we're scheduled to meet Monday January 16th.

That's tomorrow or today according to whether you read this on Sunday or
not...

Seems thtat quite a few folks will be bringing voip phones with them

I think one person was going to bring a Linksys WRT54G to install on
also.

-- 

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
http://www.bybent.com                 Get Counted!
http://counter.li.org/
Perl - http://www.perl.org/           Perl User Groups -
http://www.pm.org/
Jabber:  wwalker@jabber.gnumber.com   AIM:     lwwalkerbybent
IRC:     wwalker on freenode.net

------------------------------

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From mGunter@DatabaseCity.com Mon Jan 16 12:19:51 2006
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Will see you all tonight.  Am looking to Asterisk

to provide a platform for custom IVRs, SIP and IAX2 

accounts to other IVR shops, and for a business

I'm thinking about.   I am a reseller for PointOne,

and have administrative experience with Sipura,

and Telco Systems ATAs, and Cisco phones.

 

Unfortunately none of that experience has been

very hands on.  I am more likely to be interested

in the interpersonal networking than the IP networking.

 

Mike Gunter

VoxFlow

  

  


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From lenny@VoIPing.com Tue Jan 17 16:48:42 2006
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I was sent this today from another friend, and it was fairly interesting
(since I too have seen Gil get Asterisk going on a Soekris system).  

	http://www.tomsnetworking.com/Sections-print-article153.php

It also had a link to AstLinux (an Asterisk distro); haven't tried
it but it might be useful to the first timers... (like Asterisk @ Home)

	http://www.astlinux.org/

The other interesting piece is they built a set of VMWare images,
and VMWare (www.vmware.com) supports a free VMware Player ...

Take care,
L.
-- 
Lenny Tropiano                          E-mail: lenny@voiping.com
Partner, Networking Specialist          Pager:  pager-lenny@voiping.com
VoIPing, LLC                            URL:    http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-0867   Mobile: 512-698-VOIP [8647] 
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I also want to thank everyone for the excellent meeting last night.
The attendees were an interesting an interesting mix of newbies, experts 
(a surprising number) and those like me in the middle.

Larry


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Subject: [A*UG] Help, I'm trapped in a NAT and can't get out.
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I've been having getting my AAH system working
and thought all problems were solved last night when
Chris got my Grandstream working.

Printed out all the Grandstream settings and tried the phone as soon as 
possible but it wouldn't register.

Finally, I found on a web site that says, if you are behind a NAT,
you have to do something.  Like Stun - whatever the heck that is.
Perhaps some kind of proxy would work.

Does anyone know a quick and easy way out of the NAT trap?
If it has to be the hard way, what to do?

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From fc4linux@kojetin.org Wed Jan 18 01:44:59 2006
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Subject: RE: [A*UG] Help, I'm trapped in a NAT and can't get out.
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How do you like the Grandstream phones? I have been using the soft phones
for awhile now and can't seem to get the nerve to dish out the money for an
actual IP phone. I want to buy something that is of good quality but I don't
want to spend more then $100.00. 

-----Original Message-----
From: austin-asterisk-users-group-bounces@bybent.com
[mailto:austin-asterisk-users-group-bounces@bybent.com] On Behalf Of Larry
Alkoff
Sent: Tuesday, January 17, 2006 9:29 PM
To: Austin-asterisk-users
Subject: [A*UG] Help, I'm trapped in a NAT and can't get out.

I've been having getting my AAH system working
and thought all problems were solved last night when
Chris got my Grandstream working.

Printed out all the Grandstream settings and tried the phone as soon as 
possible but it wouldn't register.

Finally, I found on a web site that says, if you are behind a NAT,
you have to do something.  Like Stun - whatever the heck that is.
Perhaps some kind of proxy would work.

Does anyone know a quick and easy way out of the NAT trap?
If it has to be the hard way, what to do?

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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AAUG Web Site: http://aaug.bybent.com/

From labradley@mindspring.com Wed Jan 18 08:42:37 2006
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I'll probably like the Grandstream phones - they are inexpensive and 
reasonably full featured.  However, because of configuration problems I 
have yet to speak over one, so have no first hand experience.

My understanding you can get the Grandstread GXP2000 for about $100.
You might also consider a solution using cheap POTS phones.

Larry

Michael Kojetin wrote:
> How do you like the Grandstream phones? I have been using the soft phones
> for awhile now and can't seem to get the nerve to dish out the money for an
> actual IP phone. I want to buy something that is of good quality but I don't
> want to spend more then $100.00. 
> 
> -----Original Message-----
> From: austin-asterisk-users-group-bounces@bybent.com
> [mailto:austin-asterisk-users-group-bounces@bybent.com] On Behalf Of Larry
> Alkoff
> Sent: Tuesday, January 17, 2006 9:29 PM
> To: Austin-asterisk-users
> Subject: [A*UG] Help, I'm trapped in a NAT and can't get out.
> 
> I've been having getting my AAH system working
> and thought all problems were solved last night when
> Chris got my Grandstream working.
> 
> Printed out all the Grandstream settings and tried the phone as soon as 
> possible but it wouldn't register.
> 
> Finally, I found on a web site that says, if you are behind a NAT,
> you have to do something.  Like Stun - whatever the heck that is.
> Perhaps some kind of proxy would work.
> 
> Does anyone know a quick and easy way out of the NAT trap?
> If it has to be the hard way, what to do?
> 
> Larry
> 


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
From Greg@towerz.com Wed Jan 18 20:19:00 2006
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I have been using the Atcom AT-320 phones. They work really well and use
the IAX2 protocol which penetrates firewalls without any problems. They
cost about $65.00 for phones and about $50.00 for their ATA AG-168V.

You can find info at www.atcom.cn  they are also known as GNet.

You can buy them on E-Bay or other places.

If you like, you can flash them to run on SIP as well.

I will bring one to the next meeting. 

Greg Shapiro
RF Services Inc.
7301 RR 620 North
Bldg. 155 Suite 279
Austin Texas 78726
Greg@BroadcastEngineer.com
512-692-9225

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Subject: [A*UG] Zultys ZIP phones
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Is anyone using a Zultys?  I've got a ZIP2x2 in hand for review, but
setting it up is making me crazy.



-- 
"Everybody can be great because anybody can serve. You don't have to
have college degree to serve. You don't have to make your subject and
verb agree to serve. You only need a heart full of grace. A soul
generated by love."  MLK

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On Wed, Jan 18, 2006 at 08:54:00PM -0600, Joe Barr wrote:
> Is anyone using a Zultys?  I've got a ZIP2x2 in hand for review, but
> setting it up is making me crazy.

Where I work we tried these phones and they're not good at all.
They are missing caller-ID name and number (I believe it only
has name), and there is no digitmap so all the dialing has to
work via a timeout.

Zultys tech support is a nightmare.

I recommend against these phones.  You may get it to work, and
it may be interesting to play with, but I wouldn't use them
for anything serious.

---
Gil Kloepfer
aaug@kloepfer.org
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How about IAX2? Is that the same protocol mentioned at the meeting that is
"great when it works", but a new or buggy version could send you into debug mode..?

John Griessen

Greg Shapiro wrote:
> I have been using the Atcom AT-320 phones. They work really well and use
> the IAX2 protocol which penetrates firewalls without any problems. They
> cost about $65.00 for phones and about $50.00 for their ATA AG-168V.
>

From chris@tooley.com Thu Jan 19 13:20:29 2006
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I just wanted to assure everyone that the next meeting will be better
prepared.  I have already arranged a meeting room that can house about
35 people without cramming into the room.

There is a screen in the room for a projector and we have access to a
projector to use during the meeting.  That room will also make it easier
to provide a network that can be shared between the people at the
meeting (to set up servers and devices/endpoints).

I believe that we decided to do an install fest style meeting next
month.  So, I'll have a server, some distros on cds, Asterisk@Home, and
other useful things.  In an effort to show how some things work, we can
use server that I bring as a SIP/IAX2 service provider.

I'll see you all next month.  I hope the turn out is as great as it was
this month.

On Tue, 2006-01-17 at 21:20 -0600, Larry Alkoff wrote:
> I also want to thank everyone for the excellent meeting last night.
> The attendees were an interesting an interesting mix of newbies, experts 
> (a surprising number) and those like me in the middle.
> 
> Larry
> 


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Hi Joe, Gil,

Very interesting to see what you said about these phones from Zultys. 
I've got that one on my potential to-do list for security
evaluation..I understand that it runs a varient of Linux.

As a personal side project, I'm conducting structured security
analysis on about a dozen SIP voip wireless phones (Cisco, Hitachi,
Senao, ACT, UTStarcom, Clipcomm, MPM, etc.) and releasing the vulns to
the public after vendor notification and at least 30 days lead time.

http://www.security.nnov.ru/source12976.html

* [Full-disclosure] ACT P202S VoIP wireless phone multiple
undocumented ports/services
* [Full-disclosure] Clipcomm CP-100E VoIP wireless desktop phone open
debug service TCP/60023
* [Full-disclosure] MPM HP-180W VoIP wireless desktop phone
undocumented port UDP/9090
* [Full-disclosure] Senao SI-7800H VoIP wireless phone wdbrpc debug
service UDP/17185
* [Full-disclosure] Clipcomm CPW-100E VoIP wireless handset phone open
debug service TCP/60023
* [Full-disclosure] ZyXel P2000W (Version 2) VoIP wireless phone
undocumented port UDP/9090
* [Full-disclosure] Hitachi IP5000 VoIP Wifi phone multiple vulnerabilities
* [Full-disclosure] Zyxel P2000W (Version1) VoIP Wifi phone multiple
vulnerabilties
* [Full-disclosure] Senao SI-680H VoIP Wifi phone undocumented open port
* [Full-disclosure] UTstarcom F1000 VoIP Wifi phone multiple vulnerabilities

If you guys (and other folks out there) have concerns about your VoIP
phone's security...and IMO you should be concerned...or other VoIP
gear - ATAs, PBX, etc...and are interested in real world security
testing and analysis of these devices please contact me.

Thanks!
--scm

From labradley@mindspring.com Tue Jan 24 18:42:51 2006
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I recently setup a hosting account at DreamHost for my two domains.
that are currently with domainmonger.

What kind of plan would I use to have of the domains as a fixed IP for 
my Asterisk server to avoid NAT problems?

I don't need a web or email account - yet <g>.

I'm new to this and have never before messed with a domain or fixed IP.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Although my AAH is starting to come alive very nicely, I cannot hear 
sound from the remote end when I make a sip call, although they hear me 
just fine.

I have udp ports 10000:20000 and 5060:5082 forwarded to the internal 
fixed ip address of the AAH box and nat=yes in sip.conf.
The 3 (so far) sip phones and AAH are natted on the same subnet.

Apparently I'm missing something.  Any idea what to look at next?

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Date: Fri, 27 Jan 2006 08:49:11 -0600
From: Gil Kloepfer <aaug@kloepfer.org>
To: Larry Alkoff <labradley@mindspring.com>
Subject: Re: [A*UG] No remote sound from originated sip call
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On Fri, Jan 27, 2006 at 04:19:36AM -0600, Larry Alkoff wrote:
> I have udp ports 10000:20000 and 5060:5082 forwarded to the internal 
> fixed ip address of the AAH box and nat=yes in sip.conf.
> The 3 (so far) sip phones and AAH are natted on the same subnet.
> 
> Apparently I'm missing something.  Any idea what to look at next?

Make sure that the phone and the Asterisk (AAH) system is actually
using 10000-20000/udp for audio streaming.  While these are de-facto
standards for this, not all devices are configured this way by
default (Polycom phones as one example).

The file that has this in Asterisk is rtp.conf (rtpstart/rtpend).

On the phone side, the configuration is entirely dependent on the phone.

You didn't say what it was you were communicating with on the remote
end.  I would definitely check rtp.conf on your Asterisk system though.

Hope that helps.

---
Gil Kloepfer
aaug@kloepfer.org
From labradley@mindspring.com Fri Jan 27 10:18:25 2006
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Gil Kloepfer wrote:
> On Fri, Jan 27, 2006 at 04:19:36AM -0600, Larry Alkoff wrote:
> 
>>I have udp ports 10000:20000 and 5060:5082 forwarded to the internal 
>>fixed ip address of the AAH box and nat=yes in sip.conf.
>>The 3 (so far) sip phones and AAH are natted on the same subnet.
>>
>>Apparently I'm missing something.  Any idea what to look at next?
> 
> 
> Make sure that the phone and the Asterisk (AAH) system is actually
> using 10000-20000/udp for audio streaming.  While these are de-facto
> standards for this, not all devices are configured this way by
> default (Polycom phones as one example).
> 
> The file that has this in Asterisk is rtp.conf (rtpstart/rtpend).
> 
> On the phone side, the configuration is entirely dependent on the phone.
> 
> You didn't say what it was you were communicating with on the remote
> end.  I would definitely check rtp.conf on your Asterisk system though.
> 
> Hope that helps.
> 
> ---
> Gil Kloepfer
> aaug@kloepfer.org
> 


Thanks for the advice Gil.

Checking rtp.conf - rtpstart=10000 and rtpend=20000 as expected.

I first communicated with my wife's cell phone.  Her phone has a Austin 
number but she was in Daytona Fl at the time and she could hear me but I 
could not hear her except for her "hello" after which her sound cut out.

Now for testing I simply call the Daytona pots number which is forwarded 
to my Austin line.  It's very convenient since I can work with both 
phones at once.  Same deal, I can hear myself talking to the sip phone 
on the pots phone but cannot hear me talking to the pots phone on the 
sip phone.

Larry


-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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When I use the alt-F9 console in Asterisk@Home I see "funny" characters 
on the screen.  Those characters are only on blue and red letters which 
seem to be when the scrolling progress messages refer to constants.

The "funnies" look kinda like Russian characters.

I'd like to know what the system is trying to tell me about progress.

I notice that $TERM=en_US.UTF-8 on the AAH Cent-os console instead of 
the normal en-US on my normal Slackware console but I'm not sure that 
information is revelent or know where to change it.

I asked about the alt-F9 console to see scrolling progress messages on 
#Asterisk irc but nobody seems to know about it <g>

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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Larry Alkoff wrote:
> Gil Kloepfer wrote:
> 
>> On Fri, Jan 27, 2006 at 04:19:36AM -0600, Larry Alkoff wrote:
>>
>>> I have udp ports 10000:20000 and 5060:5082 forwarded to the internal 
>>> fixed ip address of the AAH box and nat=yes in sip.conf.
>>> The 3 (so far) sip phones and AAH are natted on the same subnet.
>>>
>>> Apparently I'm missing something.  Any idea what to look at next?
>>
>>
>>
>> Make sure that the phone and the Asterisk (AAH) system is actually
>> using 10000-20000/udp for audio streaming.  While these are de-facto
>> standards for this, not all devices are configured this way by
>> default (Polycom phones as one example).
>>
>> The file that has this in Asterisk is rtp.conf (rtpstart/rtpend).
>>
>> On the phone side, the configuration is entirely dependent on the phone.
>>
>> You didn't say what it was you were communicating with on the remote
>> end.  I would definitely check rtp.conf on your Asterisk system though.
>>
>> Hope that helps.
>>
>> ---
>> Gil Kloepfer
>> aaug@kloepfer.org
>>
> 
> 
> Thanks for the advice Gil.
> 
> Checking rtp.conf - rtpstart=10000 and rtpend=20000 as expected.
> 


Problem solved and outgoing sip calls are working ok now.
It was a stupid typo in an Asterisk config file that I found by pure 
accident.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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I'd like to view the file that *411 picks up the directory information 
from.  For some reason I am unable to get a "hit" using #411 although 
the IVR comes on.

Seeing the file would give me some insight on what is going on.

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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On Sat, 2006-01-28 at 11:54 -0600, Larry Alkoff wrote:
> I'd like to view the file that *411 picks up the directory information 
> from.  For some reason I am unable to get a "hit" using #411 although 
> the IVR comes on.
> 
The built in directory application reads voicemail.conf

> Seeing the file would give me some insight on what is going on.
> 
> Larry
> 
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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Chris Tooley wrote:
> On Sat, 2006-01-28 at 11:54 -0600, Larry Alkoff wrote:
> 
>>I'd like to view the file that *411 picks up the directory information 
>>from.  For some reason I am unable to get a "hit" using #411 although 
>>the IVR comes on.
>>
> 
> The built in directory application reads voicemail.conf
> 
> 
>>Seeing the file would give me some insight on what is going on.
>>
>>Larry
>>

Thanks Chris.  That's what I thought but for some reason I cannot
get *411 to accept any variation of the name.

What "last 3 letters" would I dial based on the Larry Office line below?

This is my voicemail.conf:

[root@asterisk1 asterisk]# cat voicemail.conf
[general]
#include vm_general.inc
#include vm_email.inc
[default]

120 => ,Larry Office,,,attach=no|saycid=no|envelope=no|delete=no
[root@asterisk1 asterisk]#

Larry

-- 
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Slackware Linux
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> This is my voicemail.conf:
> 
> [root@asterisk1 asterisk]# cat voicemail.conf
> [general]
> #include vm_general.inc
> #include vm_email.inc
> [default]
> 
> 120 => ,Larry Office,,,attach=no|saycid=no|envelope=no|delete=no

Try placing a PIN as the first field.

eg:
120 => 1234,Larry Office,,,attach=no|saycid=no|envelope=no|delete=no
       ^^^^

Might be some weird issue of matching.

Does it accept any sort of DTMF on other IVR related menus?

If not, might try changing the dtmfmode=XXXX in sip.conf.

;dtmfmode=info			; either RFC2833 or INFO for the BudgeTone
;dtmfmode=inband		; Choices are inband, rfc2833, or info
;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
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When and where is the next meeting of the austin-asterisk-users-group?

 

Stephan Jenkins

Applied Mechanical

Office: (512) 266-7259 x151

Cell:    (512) 297-0312

Fax:    (512) 952-7855



CONFIDENTIALITY NOTICE This document and any included attachments are
Applied Mechanical confidential.  These materials are for the use of the
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<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>When and where is the next =
meeting
of the </span></font><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt;font-family:"Courier =
New"'>austin-asterisk-users-group?<o:p></o:p></span></font></p>

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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Applied Mechanical<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Office: (512) 266-7259 x151<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Cell:&nbsp;&nbsp;&nbsp; (512) =
297-0312<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
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952-7855<o:p></o:p></span></font></p>

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<p class=3DMsoNormal><font size=3D1 color=3Dred face=3D"Times New =
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any included attachments are Applied Mechanical confidential.&nbsp; =
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materials are for the use of the addressee(s) only, and are not to be =
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information may constitute inside or non-public information under
international, federal, or state laws. Unauthorized forwarding, =
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From brett-aaug@nemeroff.com Mon Jan 30 12:18:15 2006
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From: "Brett Nemeroff" <brett-aaug@nemeroff.com>
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Hi All,
I'm a local Asterisk user here in Austin. I work for a local CLEC that
utilizes both SER and Asterisk and I do VoIP Consulting/Contracting on the
side. Just writing everyone to introduce myself. Our offices are on 360
near 2244.

so what's the participation like here? Business? Individuals? Home users?
Who's out there??

When is the next meeting?

Thanks,
Brett


From Mark@Mark-S.net Mon Jan 30 16:12:58 2006
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Date: Mon, 30 Jan 2006 16:05:21 -0600 (CST)
Subject: Re: [A*UG] Hello All
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Welp, Ill take this chance to introduce myself as well. I am a high school
student who in the last year or two has picked up linux (escpecially
Gentoo Linux) and have been working with Asterisk for about 10 months.

I have done some consulting for Asterisk as well.

Just saying Hello to everyone :)


On Mon, January 30, 2006 11:37 am, Brett Nemeroff wrote:
> Hi All,
> I'm a local Asterisk user here in Austin. I work for a local CLEC that
> utilizes both SER and Asterisk and I do VoIP Consulting/Contracting on the
> side. Just writing everyone to introduce myself. Our offices are on 360
> near 2244.
>
> so what's the participation like here? Business? Individuals? Home users?
> Who's out there??
>
> When is the next meeting?
>
> Thanks,
> Brett
>
>
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
>


-- 
Mark S
mark@mark-s.net
AIM: SkramX
1.877.77.MARKS

ATTN: MarkS
PO BOX 27914
Austin, TX 78755.7914

From sjenkins@appliedmech.com Tue Jan 31 13:43:45 2006
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------=_NextPart_000_0059_01C6266C.5888ED90
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I am an asterisk admin for my company in a small environment (around 25
phones), and am looking to find some local assistance.  Our current provider
is less than responsive to service issues, and I would like to find a local
support team.  I am picking up asterisk quickly, but am reaching the point
where some guidance is needed.  Also, I have several clients on the side who
are interested in an asterisk solution.  I don't currently have enough
knowledge or manpower to support such ventures, and would like to pursue
individuals who would like to assist or partner in taking on some new
challenges and clients.

 

When and where is the next meeting scheduled?

 

Stephan Jenkins

 

Office: (512) 266-7259 x151

Cell:    (512) 297-0312

Fax:    (512) 952-7855

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am an asterisk admin for my company in a small =
environment
(around 25 phones), and am looking to find some local assistance.&nbsp; =
Our
current provider is less than responsive to service issues, and I would =
like to
find a local support team.&nbsp; I am picking up asterisk quickly, but =
am
reaching the point where some guidance is needed.&nbsp; Also, I have =
several
clients on the side who are interested in an asterisk solution.&nbsp; I =
don&#8217;t
currently have enough knowledge or manpower to support such ventures, =
and would
like to pursue individuals who would like to assist or partner in taking =
on
some new challenges and clients.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>When and where is the next meeting =
scheduled?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Stephan Jenkins<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Cell:&nbsp;&nbsp;&nbsp; (512) =
297-0312<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Fax:&nbsp;&nbsp;&nbsp; (512) =
952-7855<o:p></o:p></span></font></p>

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From wwalker@behemoth.gnumber.com Tue Jan 31 16:16:37 2006
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Date: Tue, 31 Jan 2006 16:16:35 -0600
From: Wayne Walker <wwalker@bybent.com>
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Just a heads up.

The next meeting is Monday, February 20th, 2006, at MCC hosted by
gNumber, Inc.  We've got a _much_ larger meeting room scheduled for the
rest of the year :)

The February meeting will be an Install Fest.  We'll have the space and
a good internet connection.  

I recommend doing as much install as you can before the meeting (i.e.,
the OS, and have chosen what asterisk you want to use (1.2.3, HEAD, *
@ Home, whatever).  If you haven't done this, that's OK, just leaves you
less time to get things running.

More details to follow....

--

Wayne Walker

www.unwiredbuyer.com - when you just can't be by the computer

wwalker@bybent.com                    Do you use Linux?!
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From john_g@cibolo.com Tue Jan 31 16:33:26 2006
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Stephan Jenkins wrote:

> 
> When and where is the next meeting scheduled?
> 



[jg]Meeting will likely be the third Monday of February.

From rabel@randy.abel.name Wed Feb  1 12:07:35 2006
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What perfect timing for me, that the next meeting is an install-fest.  I 
have an old, ugly PC that I'm going to be installing to, for my POC 
activities.  Once I figure it all out and get it working well, I'll get 
a new (cheap) machine to run * on. 

Regarding the install fest, I have two questions:
1) Will there be any advice and recommendations available for which VoIP 
provider to use? 
2) Anyone care to explain the different versions available?  I have 
never heard of HEAD or 1.2.3...

Randy


From brett-aaug@nemeroff.com Wed Feb  1 13:04:05 2006
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Subject: Re: [A*UG] Re: Install Fest
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Randy,
I think there will be plenty of expertise to answer these questions at the
meeting. VoIP Providers are a kind of personal decision and highly depend
on your needs.

As for the versions.. HEAD is the lastest development version that you can
check out with CVS. If you don't understand what that means, you probably
shouldn't be using it. Don't be offended by that, but if you are looking
for a production system, you shouldn't really be tinkering with
development versions anyway.

1.2.3 just happens to be a recent stable version. "stable" versions mean
that there arn't any serious issues that they presently are working on
fixing. "Stable" means that they (the developers) agree that this is the
fixed set of issues that will be addressed and nothing else. If you arn't
looking to "play" with asterisk but want to "use" it, stick with stable.
Go check out http://beta.asterisk.org/ . Look in the upper right hand
corner for stable versions.

-Brett


On Wed, February 1, 2006 12:07 pm, Randy Abel wrote:
> What perfect timing for me, that the next meeting is an install-fest.  I
> have an old, ugly PC that I'm going to be installing to, for my POC
> activities.  Once I figure it all out and get it working well, I'll get
> a new (cheap) machine to run * on.
>
> Regarding the install fest, I have two questions:
> 1) Will there be any advice and recommendations available for which VoIP
> provider to use?
> 2) Anyone care to explain the different versions available?  I have
> never heard of HEAD or 1.2.3...
>
> Randy
>
>
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
>
>


From brett-aaug@nemeroff.com Wed Feb  1 15:58:26 2006
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Hey All,
Ok, I'm trying to get involved here; So hopefully my name will look more
familiar on the list. I understand that the upcoming meeting is to be an
installfest? Is anyone suggesting any particular distros? Does anyone
consider themselves an expert on any particular distro and want to help
others use it?

As for me, I typically stick with the easier distros to manage. For me
that's:
Fedora Core 3
Cent OS 3.5/4
WBEL

I'd like to hear from anyone who regularly installs asterisk on Gentoo or
FreeBSD.

Who will be bringing the distros to be loaded? (heh, I feel like I should
be asking "Who's bringing the beer?")

How many people are we expecting to show up? How many people are new to
Asterisk? How many are new to linux? How many people are well versed in
Asterisk?

As for me, I've been using linux in production environments for about 10
years now. For everything from mail servers, web servers, asterisk, etc..
I've been doing asterisk now for.. about 3 years.. I've been in telecom
for 8 years primarily developing CLECs from the ground up. I've had the
pleasure of working with many core systems. I did a lot of fiber network
engineering (primary optical architect for Alpheus Communications/El Paso
Global Networks. Some of you might know them). Furthermore in a previous
life I also designed telephone systems for dormatories, apartment
complexes, small offices, etc..

I also did IVR/CTI using dialogic boards before all this fancy schmancy
asterisk stuff came around. Dabbled in Bayonne for a bit too (blah!).

Ok, enough tooting my own horn, I promise I'm done.. what about you? I'd
like to hear about everyone out there . Local networking can be a powerful
thing!

-Brett

From gil@kloepfer.org Wed Feb  1 23:44:50 2006
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Subject: Re: [A*UG] Installfest Distro and Background
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On Wed, Feb 01, 2006 at 03:57:52PM -0600, Brett Nemeroff wrote:
> I'd like to hear from anyone who regularly installs asterisk on Gentoo or
> FreeBSD.

I run my home PBX on FreeBSD, and have even found and fixed bugs :-)

I'm thinking about dropping the latest (now 1.2.4) onto my laptop
on FreeBSD 6.0 and I'll take it and a Sipura ATA and a phone with me
to the next meeting.  All that being said, running just Asterisk
on FreeBSD is kind of a no-brainer (except when there are bugs to fix).
It installs much like it would on a Linux box.  Zaptel isn't so easy,
and I'm not even sure I'm qualified to demo that even though I
have it working at home on a FreeBSD 5.4 system.  I've had my share
of problems with Zaptel on FreeBSD.

For those who spoke with me after the last meeting and I mentioned the
bizarre array handling hack in Asterisk:  That bizarre array handling
WAS the cause of Asterisk core-dumping on FreeBSD, and my fix has
been incorporated back into the mainline code.  That particular
hack has caught the attention of others who have replaced it in other
places already, so the code is getting better.

I still personally feel that Asterisk on FreeBSD is not ready for
driving a production environment.

Me:  I've been working with open source (in the generic sense)
applications and OSes both personally and in my career for the
past 15 years or thereabouts.  I've been a telephony enthusiast
since I was a teenager, and Asterisk and VoIP in general is kind
of a dream-come-true.  I am a systems (FreeBSD and some Linux)
and network (we have a very extensive network) admin at
UT Applied Research Labs (ARL), where I have worked for the
past 9 years.  I'm leading a project where we're replacing our
aging 1980s-vintage Nortel SL1 with Asterisk and VoIP and will
be supporting upwards of 1000 "stations" (some will be VoIP phones,
some will be analog-over-VoIP for modems, FAXes, and secure phones).
The project started about 2 years ago, and we're finally starting
to do some serious deployment throughout our site.  A year ago,
we installed redundant Asterisk systems to handle switching
between SBC, our dialup routers, the Nortel phone system, and
the new VoIP system.  Bringing all the telecom services together
has saved us quite a bit of money monthly.

I have a variety of expertise in Asterisk, including fixing some bugs
and hacking on the code a bit to solve some specific issues we had.
A*UG seems like an interesting place to discuss new ideas with
people having similar backgrounds.

Gil.
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Subject: Redundancy (was Re: [A*UG] Installfest Distro and Background)
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Gil,
I'd be interested hearing from you and others in the group about how
you've implemented redundant servers.

I've implemented redundancy using UCARP and a daemon which tests the SIP
channel using sipsak. I find that sometimes the sip channel driver dies in
asterisk but IP isn't dead so UCARP doesn't take over.

I'm not real happy with this setup. It's complicated and difficult to manage.

What have you implemented?
-Brett

On Wed, February 1, 2006 11:20 pm, Gil Kloepfer wrote:
> A year ago,
> we installed redundant Asterisk systems to handle switching
> between SBC, our dialup routers, the Nortel phone system, and
> the new VoIP system.



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Brett,
	We implemented SIP testing by placing phone calls and using "magic"
sequences of DTMF to test the call.  Unfortunately what we implemented
was trickier than it needed to be and we abandoned that set of servers
anyway (for unrelated reasons).  In our current environment I've been
considering implementing something similar but with a simpler set of
checks.

In this way we can test the channel, the provider, the provider's
ability to provide proper DTMF, and our server's availability.

Chris

On Thu, 2006-02-02 at 09:02 -0600, Brett Nemeroff wrote:
> Gil,
> I'd be interested hearing from you and others in the group about how
> you've implemented redundant servers.
> 
> I've implemented redundancy using UCARP and a daemon which tests the SIP
> channel using sipsak. I find that sometimes the sip channel driver dies in
> asterisk but IP isn't dead so UCARP doesn't take over.
> 
> I'm not real happy with this setup. It's complicated and difficult to manage.
> 
> What have you implemented?
> -Brett
> 
> On Wed, February 1, 2006 11:20 pm, Gil Kloepfer wrote:
> > A year ago,
> > we installed redundant Asterisk systems to handle switching
> > between SBC, our dialup routers, the Nortel phone system, and
> > the new VoIP system.
> 
> 
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

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Subject: [A*UG] Re: Install Fest Distros
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I'm planning on bringing my old ugly pc to the install fest with a fresh 
copy of SuSE 10.0 installed on it.  Can anyone recommend against SuSE?  
If not, I'll also bring the SuSE 10.0 DVD if anyone else wants to 
install from it as well.

Randy


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Subject: [A*UG] Installfest Distro and background
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Hey All,
Ok, I'm trying to get involved here; So hopefully my name will look more
familiar on the list. I understand that the upcoming meeting is to be an
installfest? Is anyone suggesting any particular distros? Does anyone
consider themselves an expert on any particular distro and want to help
others use it?

As for me, I typically stick with the easier distros to manage. For me
that's:
Fedora Core 3
Cent OS 3.5/4
WBEL

I'd like to hear from anyone who regularly installs asterisk on Gentoo or
FreeBSD.

Who will be bringing the distros to be loaded? (heh, I feel like I should
be asking "Who's bringing the beer?")

How many people are we expecting to show up? How many people are new to
Asterisk? How many are new to linux? How many people are well versed in
Asterisk?

As for me, I've been using linux in production environments for about 10
years now. For everything from mail servers, web servers, asterisk, etc..
I've been doing asterisk now for.. about 3 years.. I've been in telecom
for 8 years primarily developing CLECs from the ground up. I've had the
pleasure of working with many core systems. I did a lot of fiber network
engineering (primary optical architect for Alpheus Communications/El Paso
Global Networks. Some of you might know them). Furthermore in a previous
life I also designed telephone systems for dormatories, apartment
complexes, small offices, etc..

I also did IVR/CTI using dialogic boards before all this fancy schmancy
asterisk stuff came around. Dabbled in Bayonne for a bit too (blah!).

Ok, enough tooting my own horn, I promise I'm done.. what about you? I'd
like to hear about everyone out there . Local networking can be a powerful
thing!

-Brett


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All,
When and where is the next meeting? Who will be speaking, if anyone? I
looked at the info on mailing list website, but it's for the Jan. Meeting.
Sorry if I missed this information somewhere.

Thanks,
Brett


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Subject: Re: [A*UG] Next meeting
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February 20th, 6:30 PM
MCC Building
3925 W Braker Ln

The meeting is an install fest.

On Mon, 2006-02-06 at 07:47 -0600, Brett Nemeroff wrote:
> All,
> When and where is the next meeting? Who will be speaking, if anyone? I
> looked at the info on mailing list website, but it's for the Jan. Meeting.
> Sorry if I missed this information somewhere.
> 
> Thanks,
> Brett
> 
> 
> _______________________________________________
> Austin-Asterisk-Users-Group mailing list
> Austin-Asterisk-Users-Group@bybent.com
> http://buzzard.onr.com/mailman/listinfo/austin-asterisk-users-group
> AAUG Web Site: http://aaug.bybent.com/
-- 
Chris Tooley
512-646-1507
ctooley@gnumber.com

From n5xwb@hal-pc.org Sat Feb 11 19:40:52 2006
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 From a friend in England.  Has anyone played with this system?

-------- Original Messa
