Re: [on-asterisk] IT360.ca invitation to independent Asterisk User Group members

2008-03-17 Thread McQuiggan, Mark - Broadridge (Toronto)
Simon:

I have a coworker (who is not yet a TAUG subscriber) who is very
interested in this offer.  Can he take advantage of this offer?

Thanks,

Mark.
-Original Message-
From: Simon P. Ditner [EMAIL PROTECTED]
To: asterisk@uc.org, [EMAIL PROTECTED]
Subject: [on-asterisk] IT360.ca invitation to independent Asterisk User
Group members
Date: Sun, 16 Mar 2008 22:56:35 -0400 (EDT)


Perhaps the word 'developer' scared some of you off? We mean this to be
for independent TAUG members in the sense that you don't have some big
company backing you. No need to be a C developer ;-)

So, first, there is the FREE trade show pass, with code TS1, which gets
you into the keynotes and includes the talks by Mark Spencer and Kevin
Fleming, and the TAUG meeting in the evening is also free.

If you are interested in the full conference and you fit in the category
of an independent asterisk user group member, the 60% discount code
applies to you, you would choose the Asterisk  Open Telephony
Conference at https://www.exporeg.com/it360/, and use the code HS60,
which gets you the $380+gst price until Mar 17, $420+gst afterwards.

Otherwise the 25% discount code applies, A101, which is $712.50+gst until
Mar 17 and $787.50+gst afterwards.

Cheers,
spd


~~

Your Invitation to SAVE $570 before March 17
=

This is your invitation to attend the Asterisk  Open Source Telephony
Conference sponsored by TAUG and Digium Inc. We are able to offer you
savings of $570 to attend if you register by March 17th. You pay only
$380. After March 17th, save $630 and pay only $420.

Use code HS60 or contact me (Simon Ditner) for further information.

Free Digium Keynotes for Asterisk User Group Members


There are eight keynote presentations during IT360 which are all
automatically included with your registration. Two of the eight are Mark
Spencer and Kevin Fleming from Digium Inc. Read on for their presentation
details:


KEVIN FLEMING ~ DIRECTOR OF SOFTWARE DEVELOPMENT, DIGIUM INC.
April 8, 2008 : 3:00pm - 4:00pm

Asterisk Virtualization

Abstract: This session will discuss the benefits, pitfalls and
economic advantages of running Asterisk on popular virtualization
environments including VMware and Xen. Virtualization provides virtual
PBX services, and support complex redundancy and failover models. It is
valuable for geographic distribution of services with relatively simple
reconfiguration when network changes are required.


MARK SPENCER ~ FOUNDER  CTO, DIGIUM INC.
April 9, 2008 - 12:10pm - 1:00pm

The Future of Open Source Telephony

Abstract: Mark Spencer, Asterisk creator, will present the The Future
of Telephony, including new and exciting evidence of Asterisk's growing
business presence around the world.


Summary of Asterisk  Open Telephony Conference
April 7-9, 2008
===

April 7, 2008

1) A Gentle Intro to VoIP and SIP for Business ~ Henry Coleman
2) Integrating Asterisk: Round Table Forum ~ Jim Van Meggelen, Kevin
   Broadfoot, Leif Madsen, Simon Ditner

 April 8, 2008

1) Asterisk as an Appliance: Plug  Play Telephony ~ Kristian Kielhofner
2) Asterisk Manager: Interface (AMI): When, Why and How ~ Clod Patry
3) Why Voice Matters in a Web 2.0 World ~ Matthew Gamble
4) Asterisk Virtualization ~ Kevin Fleming (Keynote)
5) Clustering Asterisk ~ Leif Madsen

 April 9, 2008

1) Migrating from POTS to VoIP ~ Jared Smith
2) Open Source Call Centres ~ Matt Florel, Reza M. Reza
3) The Future of Open Source Telephony ~ Mark Spencer (Keynote)
4) Building Blocks for Powerful Phone Applications ~ Simon Ditner
5) Case Studies: Large Business Use of Asterisk ~ Jim Van Meggelen


Links With More Info


1) Official Asterisk  Open Telephony page:
http://www.it360.ca/2008/asterisk.cfm

2) Full Conference Summary (90+ sessions):
http://www.it360.ca/2008/conf_glance_day.cfm

3) Registration Page: https://www.exporeg.com/it360/


QUESTIONS?
==

Contact Sharon Vernon, guest services.
905-695-0123 x214
[EMAIL PROTECTED]

Event Website: http://www.it360.ca/

REGISTER HERE: https://www.exporeg.com/it360/

++ Remember to contact me for your code: [EMAIL PROTECTED]






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[on-asterisk] Smallest Asterisk Installation

2008-03-17 Thread Don Moskaluk
I need a few opinions.

 

I came across this product and I started to scratch my head.
http://www.picotux.com/indexe.html 

What is the smallest footprint that Asterisk could run on?  Do you think
that 5 -10 extensions could be run on this device?

 

Don Moskaluk



RE: [on-asterisk] Smallest Asterisk Installation

2008-03-17 Thread Jim Van Meggelen
Don Moskaluk wrote:
 I need a few opinions.
 
 
 
 I came across this product and I started to scratch my head.
 http://www.picotux.com/indexe.html
 
 What is the smallest footprint that Asterisk could run on?
 Do you think that 5 -10 extensions could be run on this device?

With Asterisk, what you need to engineer for is number of simultaneous
calls, not number of registered devices. I would say that anything with more
than 200MHz worth of CPU could be counted on to handle at least 10 calls
with no troubles, as long as transcoding and echo cancellation were not
involved.

Another question I'd have is why this device?. This thing costs more than
a Linksys WRT54GL, and while I personally love tiny computers, at $100 per,
one has to understand the value of the miniature form factor in terms of the
problem being solved.

Jim


--
Jim Van Meggelen
[EMAIL PROTECTED]
http://www.oreillynet.com/pub/au/2177

A child is the ultimate startup, and I have three. 
This makes me rich.
Guy Kawasaki
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Re: [on-asterisk] Smallest Asterisk Installation

2008-03-17 Thread Ovidiu Sas

Jim Van Meggelen wrote:

Don Moskaluk wrote:

I need a few opinions.

I came across this product and I started to scratch my head.
http://www.picotux.com/indexe.html

What is the smallest footprint that Asterisk could run on?
Do you think that 5 -10 extensions could be run on this device?


With Asterisk, what you need to engineer for is number of simultaneous
calls, not number of registered devices. I would say that anything with more
than 200MHz worth of CPU could be counted on to handle at least 10 calls
with no troubles, as long as transcoding and echo cancellation were not
involved.

Another question I'd have is why this device?. This thing costs more than
a Linksys WRT54GL, and while I personally love tiny computers, at $100 per,
one has to understand the value of the miniature form factor in terms of the
problem being solved.

If you slim down the config, you will be able to fit asterisk in about 
6M of RAM (SIP only).  At 266MHz (ARM) you will be able to handle around 
10 calls until the idle CPU will bounce between 0 and 30% idle.
Check out the linux Linksys routers (as Jim mentioned) or the Linksys 
NSLU2 device (more RAM) - http://www.nslu2-linux.org/


For most of the hackable routers and NAS, you can find precompiled 
packages under the optware feeds.



Regards,
Ovidiu Sas

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[on-asterisk] Consumer grade IP Phone Set or reliable ATA

2008-03-17 Thread Bruce Nik

Good Evening All :)

For those of you who have experience in the area and have actually implemented 
large volume of IP Phone sets or ATAs I have a few questions. These units will 
be setup to connect to one central system (Asterisk) from outside. So, NAT is 
probably important. The following are the requirements that have to be met by 
the IP Phone set or the ATA:

IP Phone:
-Support SIP or IAX (MGCP support is a bonus)
-Have at least 1 X LAN in addition to the 1 X WAN port 
-Support G.729 besides supporting G.711 (I can settle for just G.711 if I have 
to)
-NAT friendly (Probably this rules out lots of sets)
-Cheapest available and looking for any or all options available. (color or no 
style don't bother me) 
-Provisionable over HTTP or TFT (this would be one amazing feature that
I would love to have on the ATA or the phone; No necessary feature)

ATA
-Support SIP or IAX (MGCP support is a bonus)
-Have at least 1 X LAN in addition to the 1 X WAN port
-At least 1 X FXS port (Don't really need FXO but if it's available I don't 
care; 2 X FXS would be a bonus)
-Support G.729 and G.711 (can settle for G.711 if have to)
-NAT friendly
-Provisionable over HTTP or TFT (this would be one amazing feature that I would 
love to have on the ATA or the phone; No necessary feature)
-Cheapest available and looking for any or all options available.

I know that Linksys allows provisioning but I think even their program is hard 
to obtain or at least costs a lot. Please let me know if you know otherwise.
Since lots and lots of these units are needed so I don't think that price or 
provisioning would be a problem. Please remember to definitely keep prices 
under $100.

Your input is much appreciated,
Bruce

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[on-asterisk] Bell strike likely imminent

2008-03-17 Thread Bill Sandiford
If you, like myself, do a lot of business with Bell Canada on a regular basis 
you may want to have a read of the following.

Bell's new release:
http://www.newswire.ca/en/releases/archive/March2008/17/c4822.html

The union's new release:
http://www.newswire.ca/en/releases/archive/March2008/17/c4803.html

The Union's memo to members:
http://www.cep25.com/documents/Vote%20results.pdf

The issues from a union perspective:
http://www.cep25.com/documents/bell_summary_offer_080131.pdf

So it looks like we are headed for some tough weeks ahead.

Bill Sandiford
Telnet Communications
905-674-2000 x100
[EMAIL PROTECTED]

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RE: [on-asterisk] Bell strike likely imminent

2008-03-17 Thread Bruce Nik

With BTS not being part of the possible strike Bell will probably have the 
upper hand. CO managers and BTS employees should be able to cover the field and 
the CO. There might not be any real issues, probably not at CLEC level at 
least. 
 
Bruce From: [EMAIL PROTECTED] To: asterisk@uc.org Date: Mon, 17 Mar 2008 
21:36:37 -0400 Subject: [on-asterisk] Bell strike likely imminent  If you, 
like myself, do a lot of business with Bell Canada on a regular basis you may 
want to have a read of the following.  Bell's new release: 
http://www.newswire.ca/en/releases/archive/March2008/17/c4822.html  The 
union's new release: 
http://www.newswire.ca/en/releases/archive/March2008/17/c4803.html  The 
Union's memo to members: http://www.cep25.com/documents/Vote%20results.pdf  
The issues from a union perspective: 
http://www.cep25.com/documents/bell_summary_offer_080131.pdf  So it looks 
like we are headed for some tough weeks ahead.  Bill Sandiford Telnet 
Communications 905-674-2000 x100 [EMAIL PROTECTED]  IMPORTANT NOTICE: This 
message is intended only for the use of the individual or entity to which it is 
addressed, and may contain information that is privileged, confidential and 
exempt from disclosure under applicable law. If the reader of this message is 
not the intended recipient, you are hereby notified that any dissemination, 
distribution or copying of this communication is strictly prohibited. If you 
have received this communication in error, please notify the sender immediately 
by email and delete the message. Thank you.
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[on-asterisk] Support services needed.

2008-03-17 Thread Kevin Anderson
I have a Trixbox install that works fine for the softphones on IAX.

I now have some hard phones that I'm attempting to connect through PAP2 
adapters.

I've
got them connected.  They can successfully dial a number, but there is
no sound heard in either direction.  I BELIEVE I have all the
appropriate ports open.  There is a NAT at each end of this connection.

I'm glad to pay someone for assistance for the last bit of config here.  Please 
contact me offline with a quote.

Thanks in advance.
Kev.

RE: [on-asterisk] Support services needed.

2008-03-17 Thread Chuck Mariotti
I would suggest trying a SIP softphone first. Easier to debug with Log Files, 
etc...

-Original Message-
From: Kevin Anderson [mailto:[EMAIL PROTECTED]
Sent: March-17-08 10:56 PM
To: asterisk@uc.org
Subject: [on-asterisk] Support services needed.

I have a Trixbox install that works fine for the softphones on IAX.

I now have some hard phones that I'm attempting to connect through PAP2 
adapters.

I've
got them connected.  They can successfully dial a number, but there is
no sound heard in either direction.  I BELIEVE I have all the
appropriate ports open.  There is a NAT at each end of this connection.

I'm glad to pay someone for assistance for the last bit of config here.  Please 
contact me offline with a quote.

Thanks in advance.
Kev.

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[on-asterisk] RE: [biz] Consumer grade IP Phone Set or reliable ATA

2008-03-17 Thread Bruce Nik

I have not heard good things about NAT support on polycoms and also they are 
really pricey for this project unless you know of a place where they can be 
found for less than $50. Date: Mon, 17 Mar 2008 22:43:08 -0400 From: [EMAIL 
PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [biz] Consumer grade IP Phone 
Set or reliable ATA CC: asterisk@uc.org; [EMAIL PROTECTED]  Bruce,  In my 
several years experience working at www.VoIPGizmos.ca - most of the knowing 
developers and implementors buy Polycom 330s or 501s to facilitate the 
implementation, deliver great return on investment as well as rely on easy use 
of the phones.  --  Alex Kovalenko Director of Operations 
http://www.kovasys.com [EMAIL PROTECTED] Tel: 888.568.2747 x701  IT 
Consulting  Staffing Consultation et Recrutement en TIOn Mon, Mar 
17, 2008 at 9:09 PM, Bruce Nik [EMAIL PROTECTED] wrote:   Good Evening 
All :)   For those of you who have experience in the area and have actually 
implemented large volume of IP Phone sets or ATAs I have a few questions. These 
units will be setup to connect to one central system (Asterisk) from outside. 
So, NAT is probably important. The following are the requirements that have to 
be met by the IP Phone set or the ATA:   IP Phone:  -Support SIP or IAX 
(MGCP support is a bonus)  -Have at least 1 X LAN in addition to the 1 X WAN 
port  -Support G.729 besides supporting G.711 (I can settle for just G.711 if 
I have to)  -NAT friendly (Probably this rules out lots of sets)  -Cheapest 
available and looking for any or all options available. (color or no style 
don't bother me)  -Provisionable over HTTP or TFT (this would be one amazing 
feature that  I would love to have on the ATA or the phone; No necessary 
feature)   ATA  -Support SIP or IAX (MGCP support is a bonus)  -Have at 
least 1 X LAN in addition to the 1 X WAN port  -At least 1 X FXS port (Don't 
really need FXO but if it's available I don't care; 2 X FXS would be a bonus) 
 -Support G.729 and G.711 (can settle for G.711 if have to)  -NAT friendly 
 -Provisionable over HTTP or TFT (this would be one amazing feature that I 
would love to have on the ATA or the phone; No necessary feature)  -Cheapest 
available and looking for any or all options available.   I know that 
Linksys allows provisioning but I think even their program is hard to obtain or 
at least costs a lot. Please let me know if you know otherwise.  Since lots 
and lots of these units are needed so I don't think that price or provisioning 
would be a problem. Please remember to definitely keep prices under $100.   
Your input is much appreciated,  Bruce
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[on-asterisk] Mass Voicemail Blasting

2008-03-17 Thread Chuck Mariotti
I noticed the new version of Trixbox has Voicemail Blasting, however, it 
appears to only allow sending a single voicemail out to a number of extensions 
on your own PBX.

Is there anything available to allow a call center rep that is calling clients 
but get's their voicemail, to leave a recorded message in the voicemail box 
instead of having to verbally say everything over and over again?

Regards,

Chuck

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